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CMP

Executive PerspectivesOn IP Communications

Feature Articles

2

Advanced IP Solutions:Transforming BusinessCommunicationsJoe Braue , CMP Media

IP CommunicationsPlaybook

Introduction

From the editors of:

Sponsored by:

Optimize

Talking ConvergenceElizabeth Ussher . . . . . . . . . . . . . . . . . .3

Homework Assignment for CIOs:Unified CommunicationsOptimize Staff . . . . . . . . . . . . . . . . . . .7

Network Computing

Enterprise VoIP Solutions: Join the VoIP PartyPeter Morrissey . . . . . . . . . . . . . . . . . . .9

IP PBXes: Live in Our Labs Sean Doherty and Peter Morrissey . . .13

Prepare your Network for VoIP:Early Assessment is the KeyPeter Morrissey . . . . . . . . . . . . . . . . . .22

InformationWeek

Tax Firm DeploysUnified MessagingPaul Travis . . . . . . . . . . . . . . . . . . . . .36

CommWeb

Avaya Shoots for Olympic GoldCommWeb Staff . . . . . . . . . . . . . . . . .37

Network Magazine

SIP Goes Prime TimeDavid Greenfield . . . . . . . . . . . . . . . .26

VoIP Races for the Remote OfficeDavid Greenfield . . . . . . . . . . . . . . . ..30

A Compendium of Recent Articles by CMP Editors

Advanced IP Solutions: Transforming Business Communications

Talk is tough. Ask companies crippled by outdated business communications technology. Crushing costs takea toll on the bottom line, burdensome management takes a bite out of productivity, and new initiatives forgrowing the business take a back seat to pushing PBXes and Centrex systems past their reasonable lifespans.

Enter the next generation of IP communications. Convergence is giving companies of every size the opportu-nity to shed their antiquated and inefficient technologies—and see the benefits of running voice over thesame network that carries data. Taking advantage of IP’s circuit-less connections (and its ubiquity) to trans-port voice does more than lower total cost of ownership (TCO). It eases administration, centralizes opera-tions, and helps employees take advantage of variety of applications that enhance productivity and ensurecustomer satisfaction.

But while new IP-based solutions free businesses to focus on growth, they also impose some serious chal-lenges. That’s what makes this IP Communications Playbook mandatory reading. Featuring the targeted con-tent of CMP’s industry-leading publications, it gives IT professionals a comprehensive overview of new IP-based solutions—and how they’ll impact every aspect of their business.

META Group’s Elizabeth Ussher gets the discussion under way in Optimize (see “Talking IP”). Running voiceover IP is no doubt better for the business, she says. But she also makes it clear that companies will have toput a new emphasis on strategy and business processes to ensure a successful implementation.

Next up, Optimize examines the unexpected effects of convergence (see “Homework Assignment For CIOs:Dealing With Unified Communications”). Running voice, data, and other traffic on a single network doesmore than transform the business; it could also transform the way employees contribute to the business—and how their lives outside the office can be incorporated into the work they do.

Network Computing’s Peter Morrissey brings things down to the all-important equipment level with an in-depth discussion of IP PBXes (see “Enterprise VoIP Solutions: Join The VoIP Party”). Sean Doherty then joinsMorrissey to put the devices through their paces. How did today’s leading products perform in tests? (see “IPPBXes: Live In Our Labs”).

As companies combine infrastructures, session initiation protocol (SIP) looks to be the standard that will pavethe way. But Network Magazine’s David Greenfield notes that companies may have to wait to take advantageof the cross-application features that SIP servers bring to the enterprise (see “SIP Goes Prime Time”).

Greenfield returns with a practical examination of branch-office VoIP implementations (see “VoIP Races ForThe Remote Office”). Find out how half-a-dozen VoIP vendors fared on everything from pricing to perform-ance when invited to build out a branch-office IP telephony network. Then check out what NetworkComputing’s Peter Morrissey has to say about the network design issues companies should address beforeinstalling that first IP phone (see “Prepare Your Network For VoIP Early Assessment Is The Key”)

Finally, it’s time assess some real-world implementations. Paul Travis of InformationWeek reports on GrantThorton, which is harnessing new IP applications to put traveling workers in better touch with clients and co-workers (see “Tax Firm Deploys Unified Messaging”). And CommWeb details the role of software-based IPtelephony in NBC’s broadcast of last year’s Olympics (see “Avaya Shoots For Olympic Gold”).

Advanced IP solutions help companies direct their energies (and investments) into initiatives that grow thebusiness. If you’re looking to capitalize on the new generation of convergence technologies, crack open theIP Communications Playbook.

Joseph BraueVice President/DirectorCMP Integrated Marketing Solutions

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From theeditors of:

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Optimize, August, 2004

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Talking ConvergenceThe technology may be efficient and scalable, but CIOslook for ROI and security assurances

Maturing voice-over-IP stan-dards and technology aremaking the concept of deploy-ing converged infrastructuresmuch more viable. But asCIOs investigate the ins andouts of converged infrastruc-tures, they’re learning that

much more than technology is at stake. Theircompanies’ financial statements will have animpact on their deployment decisions, and newchallenges and opportunities for managingemployees and business processes will emerge.

The Meta Group recently released an updateto our original 2001 study, which assesses driv-ers, time frames, and expectations for a con-verged enterprise. While implementationshaven’t increased significantly in the past twoyears, we found that enterprises that movedtoward a converged infrastructure have gainedexperience and confidence, and are expandingtheir initial investments into the technology.

The reason most commonly cited in thestudy for choosing convergence is either to out-fit a new branch or building or the end of life orcontract for PBX, Centrex, or messaging plat-forms. A frequent misconception is that CIOsare waiting for standards adoption. Ourresearch indicates that the majority of userswill converge when the time is right for them,regardless of the status of vendor-standardsadoption. The backlash we’ve seen from endusers in this area is the result of a lack ofprocess, expertise, and security. Additionally,inadequate ROI metrics and flimsy businesscases have impeded a rapid shift to the use of

By Elizabeth Ussher

EXECUTIVE SUMMARY Despite more mature

voice-over-IP technology, a new survey, updated

from 2001, shows greater caution among compa-

nies considering infrastructure convergence, and

recognition that financial statements will affect

deployment decisions, as well as pose new man-

agement challenges. Yet, companies that have

embarked on convergence projects have gained

confidence and are forging ahead with implemen-

tations of the technology.

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convergence technologies. An integral part of thebusiness case is identifying current costs and spend-ing levels. These must be isolated and normalized.

For example, the useful life of a PBX is very dif-ferent from that of a server. The number of spareparts stored on-site versus those that can be reliablyshipped overnight also varies by technology. A com-prehensive cost analysis must be compiled as abaseline before a cohesive ROI can be consideredviable. Our research indicates that many enterprisesassume they’ll save money by doing the math with-out doing the work to uncover true, per-unit costsincurred for the individual technology and support.

The number of respondents who say they don’t

plan to converge has increased from 7% to 15% inone year.Today, users are more cautious and requirereal, hard, case-specific numbers and proven suc-cesses to drive deployment. Implementation horrorstories run rampant—albeit some fueled by compet-ing vendors. Many customers that announced largeimplementations have backed out, stalled, or com-pletely changed their IP-telephony (IPT) deploy-ment strategies because of complexity, process, andinexperience on the part of the staff or VAR.

Daunting task Many enterprises realize that the concept of oneseamless, global network—as opposed to four or five

As our seven-year Centrex contract for

the Providence (R.I.) Public Library sys-

tem approached expiration two years

ago, we feared a big jump in our month-

ly phone costs–without new features or

services–if we signed another agree-

ment. But to bring telephony in-house, we faced a

strategic decision: Should we invest in legacy

voice technology or jump to voice-over-IP,

given its promising route to the future?

Our choice couldn’t be made lightly.

With a central library downtown plus nine

neighborhood branches, the library is a

major public institution in need of its phones.

In 2003, we had just over a million visitors,

and our reference desk answered more than

170,000 questions—at least 25% of them by phone.

In addition, we’re actually a private nonprofit that oper-

ates with both private and public funds to serve the city and

the state. While this heightens our sense of community

accountability, especially in patron service and cost control,

it also raised the stakes of VoIP success even higher.

Bringing voice services in-house was daunting, given

that my staff had taken over responsibility a month before

and was still on a steep learning curve. A VoIP implemen-

tation would add to that curve.

However, the phone cabling in our central library was

old, buried deep inside walls two- to three-feet thick and

was estimated to cost more than $30,000 to rip out and

upgrade. VoIP would let us use our data network instead.

After several rounds of evaluations, including a Centrex

renewal, we determined VoIP was the best route for us.

Three key lessons emerged from deploying VoIP: Don’t

assume anything in running voice services over a data net-

work; ensure that all stakeholders, especially the WAN

provider and all vendors, are clear on their roles in the VoIP

implementation and communicate frequently; and make

sure they’re clear on the implementation without

distractions from other assignments.

We followed a five-step approach to

deploying VoIP: assessing fully our require-

ments and network; developing and follow-

ing a comprehensive deployment plan that

delineates the public and private networks,

and includes equipment staging, program-

ming, testing, and turn-up; educating both

users and administrators on the systems fea-

tures and management; supporting the system,

including individual assignments when issues arise; and

reviewing performance for system tuning and optimization.

Since the library’s cutover to the Siemens HiPath plat-

form in September 2003, we’ve realized a number of ben-

efits. Among them, the system’s centralized administration

allows us more control and flexibility in making changes,

all done from our data-processing center. We avoided the

$30,000 cost and disruption of recabling the central library

by using our existing data network. And based on that cost

avoidance plus savings on monthly Centrex charges,

we’ve calculated a total cost of ownership over the antici-

pated 10-year life of the system, and figure it’s like getting

the last five years for free.—Mark Cataudella, director of

facilities for the Providence Public Library

Library Sings VoIP Praises—Quietly

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geographic networks—is much more practical andeasier to manage, but often organizationally impos-sible. One early IPT-adopter client, for example, is sorelieved that his network is now stable and process-ing calls properly—he didn’t bother with a preinstal-lation network audit—that the idea of adding a tele-phone-based application scares him to death.

Wholesale replacement of PBXs is extremelyrare. We also see a continuation of traditionalfederated decision models in companies with apresence in more dispersed geographies.Finding a balance is the bottom line here.

Restructuring requiredWe’ve found that enterprises are consolidating voiceand data staff, usually under a CIO. This trend hasproven to be the most difficult culturally, but also themost rewarding from a business standpoint.

CIOs are getting creative as they assimilateteams. For example, the CIO of a large munici-pality, particularly sensitive to cultural changes,put in place personal incentives through man-agement objectives that give the extra push herstaff needed.

Our research shows a converged staff creates itsown culture and will consider a variety of resolu-tions to a business problem, regardless of theincumbent technology vendor, as opposed to beingconvinced that the next release of “product X” willsolve it.As a result, converged staffs tend to be morebusiness-solutions-oriented and regard technologyas a tool to accomplish this. Cross-training hasincreased, but mostly this is done in-house.However, voice personnel, with their proximity touser-functionality requirements, will continue to

wield significant influence in IPT selec-tion and deployment.

For companies implementing thetechnology, key applications remainconsistent with the 2001 study; however,the relative importance of some appli-cations has increased.

Support of remote workers has risen inawareness, for example, while call centerapplications have dropped several points.We note an increase in the adoption oftools to support physically disparateteams—collaboration, conferencing, andmobility—and users perceive these tohave more impact on the business (seechart, at left).The trick is to deploy appli-

cations that are of obvious benefit to the user.CIOs should expect to deploy many applica-

tions, each meeting the needs of a differentdepartment or job function. For example, a niftyinventory-control application that sends a mes-sage to the telephone display when stocks ofnotepads are low may be interesting to the pro-curement team, but of no value whatsoever to aline worker in manufacturing. Conversely, a sim-ple application for time reporting—basicallyreplacing a time clock with the telephone—whichalso shows announcements, vacation accrued,and schedule changes, could prove very valuableto those employees. Keep in mind that users mustsee something more than a new telephone, partic-ularly if they’re aware of the expense.

Cost proved to be aparadox. Respondentsindicated cost savingsas the key driver forconsidering a con-verged network (seechart, p. 72). For exam-ple, 57% of respon-dents said they’ve seenmarked reductions inoperational costs, and60% cited lower circuitcosts. However, thehigh cost of equip-ment was noted as akey inhibitor. This wasfollowed closely byboth the lack of bud-get and security con-

Convergence ApplicationsAmong the companies that have implemented converged networks, the following applications are supported:

DATA: Meta Group survey of 241 respondents

% of respondents

20 40 60 80

Conferencing

Productivity

Remote access

User mobility

Networked voicemail

Distance learning

Unified messaging

Call-center applications

Spending On The RiseHow will spending on voice and data convergence in 2004 compare with 2003?

DATA: InformationWeek Research convergence suvery of 140 business-technology professionals

Somewhat more36%

Somewhat more36%

None 6%None 6%

Significantlymore 14%

Significantlymore 14%

Significantly less4%

Significantly less4%

Same31%Same31%

Somewhat less9%

Somewhat less9%

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cerns. Roughly 36% of respondents attributed anincrease in staffing costs to their converged net-work, and 44% cited an increase in infrastructurecosts. Users considering a converged network mustrespect the fact that all costs won’t decrease, andchoose the components that are most attractive totheir particular organization.

Operational-cost reduction and ROI aren’t thebottom line. Users should plan to invest for at leastfive years in adequate IT infrastructure as a foun-dation for application support, communications,and other critical business functions. A flexible,reliable, low-maintenance infrastructure withissue-free implementation, as well as the availabil-

ity of services and technical expertise, are factorsthat usually outweigh cost concerns and are moreimportant than any specific features, functions, orcost savings. Further proof of this concept can befound in the increasing number of companies tak-ing advantage of the opportunity to deploy storagearchitectures that comply with Sarbanes-Oxley Actrequirements. Spending on security also illustratesusers’ willingness to invest in infrastructure forissues more important than cost savings.

The implementation phaseWhen asked which vendor categories they’d con-sider to provide the best convergence solution, theresults were more scattered. “Converged-systemsvendor” with no further clarification ranked No. 1,but with a mere 28% as opposed to the 2001 study,which provided a 48% response. Experience hasshown that enterprises will choose best of breed.

Therefore a single vendor isn’t always ideal.Systems integrators followed with a 20% rating,

and then existing PBX vendors, with 13%. Thiscould actually be combined with telecom inter-connect at 10%, bringing the total to 23%; in mostcases, these are the same organization. This showsmore confidence in companies experienced withvoice and a PBX heritage. Data vendors and data-networking VARs made poor showings at 9% and8%, respectively.

We often observe a variety of VARs involved in aconverged implementation—one for the data infra-structure, a second for the voice application, a thirdfor wireless—each representing its own specialty,even in cases where the equipment is from the samevendor—for example, Alcatel, Cisco Systems, orNortel Networks.This is much the same as tradition-al PBX implementations, where different peripheraldevices or applications—for example, interactivevoice response or call center—required expertise ina given domain.

Most convergence ROI models are based onbringing operational activities in-house. The vendorsales process maps clearly to the study: Once a con-verged system is deployed, 61% of the respondentsexpect to transition operational activities to internalresources. Twelve percent would insource specificoperations and components, and 27% would contin-ue to outsource. However, the internal process andstructure to support telephony, conferencing, andmessaging will have to be created or fine-tuned inmany cases.

In fact, the study indicates that IT organizationswill be encouraged to insource more operationalcommunications services, while continuing to out-source specific projects and implementations.Concurrently, we expect traditional PBX vendors tocontinue to simplify operational user interfaces.This will increase the opportunity for users totransition internal management and integration oflegacy systems to internal resources, while ensur-ing that traditional customers can also exploit theseoperational efficiencies.

Convergence of technologies onto a single IPinfrastructure will change the structure of enterpriseoperations. Users must prepare for the impendingevent wherever it happens, with a business case sup-porting the decision on whether or not to converge. O

ELIZABETH USSHER is a VP at Meta Group.

Please send comments on this article to [email protected].

The Urge To MergeWhat factors are driving your company to merge its voice and data systems onto one system?

NOTE: Multiple responses allowedBASE: 100 sites converging traffic over its IP networkDATA: InformationWeek Research convergence survey of 140 business-technology professionals

% of respondents

20 40 60 80

Reduce costs

Improve productivity

Ability to provide anytime/anywhere access to corporate data

Better quality of service/applications

Better collaboration

Implementation of a universal in-box

Faster responsiveness to customers

Elimination of landlinetelecommunications

January 2005

Telecommunications providers and CIOs alikeshould pay close attention to the convergence ofpeople's work, home, and mobile worlds,according to research firm In-Stat, based on arecent survey. Driven by overlapping work andnon-work uses for certain telecom servicesacross multiple technologies, we could soon seethe development of what analyst Amy Cravenscalls "integrated residential services"—whichwould provide a single point of access for voice,data, applications, and messaging.

"It's not the convergence of voice and data orthe integration of the fixed -line and wirelessthat's important," Cravens says. "It's the conver-gence of work, home, and mobile life."

The trend is already beginning with increasedtelecommuting and voice-over-IP networks thatlet people easily forward calls to where theyare—also known as "find me" mode. Web editorHoward Baldwin talked to Cravens about howsuch a convergence would further affect CIOs.

Q: Would you characterize the convergenceyou're talking about?

A: We're talking about business calls that mightbe forwarded to a home phone. It's voice mailand E-mail integrated, so they create a moresynthesized and streamlined communicationsenvironment. Workers who have communicationsgoing on throughout the day will want to meldthem together.

Q: Does this stem, in part, from increasingglobalization? If you're on the West Coast andneed to talk to people in India, they're 12hours ahead of you. That means either callingat 8 p.m. or 5 a.m. to catch them at the begin-ning or end of their day.

A: That's an important factor, but even moreimportant is the telecommuting angle—peoplewho work from home one day a week, or anhour or two in the morning before they take thekids to school. For them, it's important to haveconverged communications with the office.

Homework AssignmentFor CIOs

Dealing with unified communications

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Q: What do CIOs have to think about to makethis work?

A: They want employees to be productive wher-ever employees might be, whether they'retelecommuting or in a wireless hotspot. Theseintegrated services take that one step further, todeliver voice mail or E-mail through a wirelessnetwork, and to do it from one source, soemployees can be fully in touch wherever theymay be.

Q: Is this the integrated voice-and-data net-work that VOIP brings?

A: Eventually, all these capabilities might be on asingle pipe. But right now, in the near term,some of these may be done across various net-works.

Q: How much more productivity can youexpect to get from employees with a systemlike this?

A: It would be hard to estimate. There have beenstudies regarding telecommuters and working athome during off hours. But nothing's been doneregarding what communications integration doesfor productivity.

Q: It seems there are multiple issues to bedealt with regarding systems managementand password protection.

A: That's true. Anytime a laptop leaves the cor-porate premises, whether to go to Starbucks orto an individual's home, there are issues. Thehome environment may be seen as more secure,as opposed to a public venue, but it's still off-premise. The computer's got to be password-protected to keep the kids from using it. There'squite a bit of work being done by serviceproviders developing computer clients thatensure that you're logging on through a VPN interms of securing communications.

It will be a while before we see these servicesoffered widely. Even more than the technicalissues for service providers is the consumer edu-cation that will need to take place, explaining thevalue and benefits of these services. Consumerscan understand the value of integrating their cellphone and their landline services; the other net-work services, such as voice and E-mail integra-tion, will be less well-understood. That will bemore of a barrier to adoption.

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Optimizemag.com, January 2005

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Network Computing, March 17, 2005

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VoIP:

PAJO

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RTYIN THE

Vendor competition, standards maturityand enhanced features make it a fine timeto be in the market for an IP PBX

BY PETER MORRISSEY

If you’ve yet to launch even a VoIP (voice over IP) pilot,

you might feel like the classic lonely guy knocking

around his empty apartment while a wild bash rages one

flight up. Only instead of missing out on loud music, iffy

shrimp cocktail and a screaming morning-after headache,

your company is doing without productivity enhancers such

as GUI-based soft phones that can integrate with desk

phones and address books. The same application may also

communicate presence information about the availability of

others on the system. In addition, a number of IP PBXs have

a “find me, follow me” feature that lets you contact a cell

phone if, for example, the office phone doesn’t answer. Some

will ring an office phone and a cell phone at the same time.

Moreover, it’s a buyer’s market for VoIP systems. Gartner

reports that at year-end 2004, sales of new IP-PBX systems

surpassed those of conventional PBXs, and that by 2010,

IP-telephony products will represent 90 percent of new

system sales.

Not long ago, Cisco Systems dominated the VoIP market,

thanks to its 1998 purchase of Selsius. Legacy providers were

34 EXECUTIVESUMMARY36 READER POLLRESULTS 39 LIVE INTHE LABS: IP PBXs42 RFP SCENARIO46 HOW WE TESTED

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thrown off guard initially, but the tide is quickly turn-ing. One Dell’Oro market share report (see chart below)shows Avaya leading Cisco in large IP PBX shipmentsin the third quarter of 2004, with Nortel Networks aclose third. Even more significant, this tally doesn’ttake into account hybrid systems from Avaya and Nor-tel that combine TDM (time-division multiplexing) andVoIP ports. There are many smaller players to choosefrom as well, including Zultys Technologies andVonexus, which both participated in our review.Although cost is cited in our reader poll as the No. 1obstacle to VoIP adoption (with the need to upgradenetworks a close second), competition shows no signsof cooling, and that’s good news for consumers.

Standard and DeliverStandards like Ethernet, IP, PoE (Power over Ethernet)and QoS (quality of service) let you plug all componentsof a VoIP system into any other vendor’s data network,which means you can shop around for the best price.

Moreover, the SIP (Session Initiation Protocol) stan-dard makes it feasible to choose different vendors’ gate-ways for access to the PSTN, with SIP trunking betweenthe gateway and the IP PBX. SIP is currently anapproved draft standard; some advanced features arestill under construction, but many vendors are findingways to make it work.

SIP also makes it possible to integrate presence applica-tions from different vendors, as well as use phones fromone with the IP PBX of another. Many vendors have resis-ted this—after all, half their revenue comes from phones.

For our review (see “Live in the Labs: IP PBXs,” page39), we required that vendors spec out third-partyphones and be prepared to demonstrate interoperabili-ty with three different vendors’ endpoints. We testedsome advanced features, and though the results weren’t

perfect, we believe SIP is a viable protocol for multiven-dor, enterprise IP-PBX phones. Despite some interoper-ability problems, we were impressed that Vonexusrequires third-party SIP phones and Zultys uses only SIPphones. We were also pleased that Avaya has an exten-sive certification process for third-party SIP support andhas demonstrated a commitment to making it work.

This is another advantage that Avaya has over its rivalCisco. Although Cisco has support for SIP trunking in itsCallManager product, you can’t buy SIP-based phonesfrom a third party that will work with Cisco’s CallMan-ager. We hope that changes. Even though Avaya has athird-party certification program, it does not sell third-party phones, although we hope that will also change inthe future. No matter how you cut it, we recommendmaking third-party SIP support a requirement.

Look, Then LeapDon’t recklessly crash the VoIP party, however—aswith any technology decision, you shouldn’t make a

0 40%30%20%10%

3Q 2004

Nortel

Cisco

Avaya

Siemens

Sphere

Alcatel

Mitel

Shoretel

Others

3Com

NEC

Large IP PBXs: Worldwide Market Share

Source: Dell’Oro Group

Executive Summary

OENTERPRISEVoIPOur “Join the Party” theme implies that you’ll have arollicking good time implementing voice over IP.Admittedly, that’s a stretch, but not as much of oneas you might think.

First, competition among vendors is fierce. Youhave your pure-play IP PBX sellers and your conven-tional PBX vendors that have expanded to includepure-IP and hybrid offerings. Any time you can getthree or four viable RFP responses, that’s a good thing.

Second, the SIP (Session Initiation Protocol)standard is nearing completion. Indeed, the PBXsand third-party endpoints in our tests proved moreinteroperable than we’d expected.

Finally, VoIP enables new features that will makeend users and telecom managers alike toot theirparty horns; these include unified messaging, sim-pler moves/adds/changes, more geographic flexibil-ity, easier integration with new applications likepresence, and soft phones that add features andhelp manage desk phones. All these should help youmake your case for the technology investment, andmaybe even a supporting network upgrade.

In “Live in the Labs: IP PBXs,” page 39, we reporton testing IP PBXs from Avaya, Vonexus and ZultysTechnologies in our Syracuse University Real-WorldLabs®. In addition, we updated our RFP on behalf offictional underwriter HaveNoFear Insurance and fac-tored that into our scoring. Avaya’s IP TelephonySolution took our Editor’s Choice award because itprovided the best interoperability, comprehensivemanagement tools and a ton of features for the price.

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move just because you think everyone is doing it (they aren’t) or because of vague ROI claims. Yes, VoIP has matured significantly and deserves serious consideration, whether as a PBX replacement,for trunks between office locations or to enable newpresence-based services. But if your old PBX is hum-ming along and supporting the com-munications needs of your business,maybe VoIP isn’t justified. In our read-er poll, 11 percent of respondents saidthey have no plans for production-level VoIP because their current PBXswork just fine.

Of course, you should still talk tosalespeople and lay the groundworkfor a VoIP pilot. Determine whetheryou have the skills in-house to man-age the new technology and deal withnew security problems. One note: The10 percent of readers with securityconcerns should check out a free, 99-page report by the National Instituteof Standards and Technology (NIST)titled “Security Considerations forVoice Over IP Systems,” available atcsrc.nist.gov/publications/nistpubs/800-58/SP800-58-final.pdf.

Before building a purchasing pro-posal, evaluate your infrastructure toensure that your network can handleVoIP (see “Prepare Your Network forVoIP,” ID# 1513f4). If you lack QoSand high availability, it could be toughgoing. In that case, a phased approachmay be in order.

Determine the optimum mix ofTDM and VoIP for your network. Forexample, VoIP could be used to trunktogether legacy PBXs. It’s also possibleto have one PBX support a mix ofVoIP and TDM phones. Some vendors,notably Cisco, will push you toward pure IP simplybecause that’s what they sell, while legacy PBX ven-dors like Avaya take a different approach, placing noreal limitations on the number of TDM phones in themix. In fact, if you own an Avaya PBX, you may beable to add VoIP functionality and start reaping thebenefits without a forklift upgrade. Many legacy ven-

dors also offer pure IP systems backed by a long trackrecord of providing solid voice services.

Prime VoIP candidates are companies with many,geographically dispersed offices. VoIP can trunk locations together over existing data WAN circuits,saving on long-distance while making more efficient

use of bandwidth and improvingintracompany communications. It alsoprovides some flexibility in PBXs loca-tions. For example, some locationscould have only VoIP phones, access-ing a PBX at another location. Keep inmind that in some cases, VoIP calls can take up more bandwidth than cor-responding circuit-switched calls,depending on the type of codec used(for more on codecs, see the “VoIPCodecs” sidebar at ID# 1512f2). In ourreader poll, about three-fourths ofthose installing VoIP weren’t surewhich codec they were using! Anotherbenefit that VoIP provides for thosewith physically dispersed locations isthe opportunity to easily consolidatemultiple vendors’ PBX systems, cuttingmaintenance costs.

Redmond’s SIP StakeMicrosoft was an early adopter of SIP;for example, with Microsoft Office 2003applications, it’s possible to pass alongpresence information with documentattachments. This functionality doesrequire at the back end Microsoft’s LCS(Live Communications Server), whichincludes SIP and presence capabilities.

And Microsoft says its new desktopinitiative, Istanbul, due for release laterthis year, will include even richer pres-ence, video and phone capabilities,offering integration with SIP phones

and IP PBXs, as well as Microsoft Desktop applications(see “Siemens Also to Support Istanbul Directly,” atwww.commweb.com/60400304). For now, Microsoft ispartnering with Alcatel, Mitel and Siemens to integratetheir VoIP phone systems more closely with Windows,and it’s working with Polycom to integrate phones andvideoconferencing solutions.

It remains to be seen how much interoperabilitythere will be between Microsoft and the rest of theworld—in the past, the company has removed SIP func-tionality from some desktop applications withoutwarning, notably Windows Messenger. Still, there aremany advantages to integrating desktop apps withcommunications services, and this is an area we’ll bekeeping an eye on.

OUR COVER PACKAGE ON ENTERPRISE VoIP AND

SIP-COMPLIANT IP PBXs CONTINUES ON PAGE 39.

One Voice On Feb. 7, the National Insti-tute of Standards and Technology (NIST),

Purdue University’s Cerias Lab, Symantec, Qualys,Alcatel, 3Com, Enterasys, Spirent Communications,Qwest and Comcast joined with TippingPoint to formthe Voice Over Internet Protocol Security Alliance(www.voipsa.org). The group says it will work with suchorganizations as the IETF and SIP (Session InitiationProtocol) Forum to publicize possible vulnerabilities inVoIP networks and ways to counter emerging threats.

FYI

Source: NETWORK COMPUTING Reader Poll, 575 respondents

Describe your current or planned VoIP implementation.

Service provider- or carrier-supported dedicated IP PBX system on service provider site

Service provider- or carrier-supported dedicated IP PBX system on company site

Company-owned and supported dedicated IP PBX

Service provider- or carrier-supported shared IP Centrex system

What are your organization’s plans for putting enterprise VoIP into production?

We’ll install production-level VoIP within the next 24 months

VoIP is in production

No plans to install production-level VoIP

READER POLL

We’ll install production-level VoIP within the next six monthsWe’ll install production-level VoIP within the next 12 months

13%32%

15%

30%

10%

9%5% 19%

67%

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IN THE LABS: IP PBXsWe brought a trio of VoIP systems in-house to evaluateSIP compatibility. Avaya’s SIP Telephony Solution got

the first-place call, thanks to its excellentinteroperability and features-to-price ratio

BY SEAN DOHERTY AND PETER MORRISSEY

LIVEWhen we introduced our fictional under-writer HaveNoFear Insurance LLC, whichprotects the makers of reality-TV showsfrom lawsuits by disgruntled or damaged

contestants, the company was seeking a basic VoIP(voice over IP) system (see “VoIP to the Rescue,” ID#1512f2). But HNF execu-tives realized that imple-menting VoIP without SIP(Session Initiation Proto-col) could expose it to arisky vendor lock-in sce-nario. So we took HNF’sfear factor in hand and sentthe first-round vendor par-ticipants an updated RFP seeking SIP-enabled IP PBXs.We upped the ante by bringing the systems into ourSyracuse University Real-World Labs® to test their sup-port for third-party SIP phones and endpoints.

Avaya, Vonexus and Zultys Technologies put theirproducts on the line for our evaluation. Alcatel, MitelNetworks Corp., Nortel Networks, ShoreTel and SiemensAG declined. We didn’t invite Cisco Systems because it

did not participate in our first call for proposals and itsCallManager doesn’t support third-party SIP endpoints.

Our tests illustrate how vendors are embracing SIP:slowly and surely. Although Avaya’s and Vonexus’ IPPBXs are chock full of features normally found in legacyTDM (time-division multiplexing) systems, these fea-

tures did not always trans-late to third-party SIPphones. For example, Avayatold us its setup, comprisingits S8300/G700 Communi-cations System and S8500Converged Communica-tions Server, supports auto-matic callbacks and distinc-

tive ring tones for SIP phones. But that wasn’t what wefound in the labs. And Vonexus could not get any third-party phone to light the MWI (message-waiting indica-tor). Zultys was able to get its Polycom phones to lightthe MWI when voicemail was pending but could not dothe same with the SNOM phones.

The two most common explanations vendors gavefor features not working were that the third-party

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Customize the results of this report card using the Interactive Report Card®, a Java applet, at www.nwc.com.

Avaya Zultys Technologies VonexusSIP Telephony MX250 Enterprise Interaction Center

FEATURESPBX (10%) 4 3 4Unified messaging (10%) 4 3 4.5PC application (5%) 4 4.5 3.5Presence (5%) 4 4.5 4INTEROPERABILITY (30%) 4 3 2.5MANAGEABILITYAdministration interface (10%) 4.5 4 4Application integration (10%) 4.5 4 4.5Telecommuting (10%) 4 4.5 3.5PRICE (5%) 4 4 3SCALABILITY (5%) 4 4.5 4TOTAL SCORE (100%) 4.10 3.63 3.53

INTEROPERABILITY rates how well vendors’ IP PBXs work with a variety of third-party SIP-compliant endpoints.PRICE score is based on our RFI scenario.

B+ B_

B_

A≥4.3, B≥3.5, C≥2.5, D≥1.5, F<1.5 A-CGRADES INCLUDE + OR – IN THEIRRANGES. TOTAL SCORES AND WEIGHTEDSCORES ARE BASED ON A SCALE OF 0-5.

SIP-Compliant IP PBXsREPORT CARDREAL-WORLD

LABS®

phone does not support that feature or that the SIPstandard does not support that feature. There is sometruth to these statements. For example, IP PBX vendorshad trouble delivering distinctive ring tones to thethird-party phones under test. Ring tones are generatedlocally by the SIP device, so theydepend on how the phone manufac-turer implements them (for a list ofthird-party phones used, see “IP PhoneChoices,” page 49).

As for SIP, it’s young and still grow-ing (see “VoIP: Join the Party” page32, for more on SIP’s progress). Forexample, the SIP standard doesn’tdetail a signal when a phone goes offhook, so we wanted to see how ven-dors would handle that. In tests, weplaced a call over the Avaya system toa Cisco IP phone we had taken offhook. The phone rang instead of gen-erating a busy signal and forwardingthe call to voicemail. However, thesame test with the Grandstream and SNOM phones onthe Avaya system sent the call to voicemail, which iswhat we’d expected. An extension of RFC 3265 willaddress this problem, and Avaya told us it will accom-modate the feature in its next release. The other ven-dors did not get back to us. But note that the presencecapabilities of soft phones in this review could identifywhen their “bound” phone was off hook and generatea SIP “notify” signal for a “busy” status.

None of the systems received a perfect score in

our interoperability tests (see “How We Tested SIP-Compliant IP PBXs,” page 46). Although Avaya’scame out on top, it had difficulties with managed-call and blind-call transfers on the Cisco phone. Zul-tys’ setup suffered from its choice of Microsoft Mes-

senger as an endpoint and for bring-ing only two third-party SIP phones.Vonexus also didn’t bring the requiredphones to test, and it got off to afalse start with an installation prob-lem stemming from insufficient HMP(Host Media Processing) resources tosatisfy our test scenarios; in addition,it didn’t include the add-on confer-encing module specified in our RFP.Finally, Vonexus was the only vendorto hazard testing using an analogphone, an Audiocode eight-port FXSgateway.

A word on HMP—this feature, builtinto Intel CPUs, enables audio process-ing without the hardware that normal-

ly performs this function. A license key must be enteredto turn HMP on, which adds one more thing toaccount for in an installation. If you find that featuressuch as conferencing inexplicably don’t work, makesure HMP is on.

Killing Me SoftlyWe haven’t figured out whether the computer willbecome the next phone or vice versa. But we are will-ing to bet that the phone will be the next killer appli-

What’s the biggest obstacle to VoIPadoption?

Security

Current PBX was purchased recently, and there is no need to consider an upgrade

Lack of reliabilityNetwork upgrade Cost

READER POLL


Lack of standardsOther

9%10%

14%26%

11%10%

20%

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cation in hardware or software, so we carefully exam-ined each vendor’s PC-based applications. We likedZultys’ MXIE best, thanks to its implementation onthe Windows desktop and good call- and presence-management features.

We downloaded MXIE directly fromZultys’ IP PBX appliance, the MX250.We could localize MXIE for emergencyservices, such as E911, and configure itwith call-handling logic, such as callforwarding and follow-me/find-merules. It supports 16 simultaneous callappearances and multiple codecs toconfigure call optimization. In addi-tion, it comes with TAPI (version 2.1)support to hook into Windows applica-tions and installs its own print-to-faxWindows driver. Although MXIE couldbind to and easily control call processing for the third-party SIP phones under test, it lacked some UM (unifiedmessaging) features, such as the ability to listen to anddelete e-mail messages over the phone, that would havemade it a killer.

Avaya’s IP Softphone R5 application and Vonexus’Interaction Client also provided a lineup of potentialkiller applications. Like MXIE, the Avaya Softphone ini-tiates and terminates calls with a click of the mouse. Italso implements a secure IM (instant messaging) clientwith server-side logging and archiving to comply with

industry regulations. The Softphone R5 also provides apanoramic view of enterprise users with presence man-agement, just like Vonexus’ Interaction Client.

Each vendor advised us of a redundant or failoveroption for high availability, and eachhad an answer for the 20 telecom-muters accessing the central officefrom a home-office location. They alsomade recommendations for basic andexecutive phones that included userlicenses (read the complete vendorresponses at ID# 1605rd2).

Avaya chose the SNOM 200, pricedat about $378 per unit, as both its basicand executive IP phone. AlthoughAvaya says it prefers its own 4602SWSIP Phone (estimated at $284 perunit), we specified a third-party SIP

phone to bang on interoperability, and the SNOM200 fit our minimum requirements of two 100-Mbpsports plus support for IEEE 802.3af and 802.1Q/p.Although a better choice than Zultys’ entry, the $200SNOM 105 (which did not meet minimum require-ments and is no longer manufactured), the SNOM200 lacks a hard key for call transfers, forwards andholds, and was also without a key for volume control.We did find all these features in the Avaya 4602SWSIP Phone and Vonexus’ basic phone choice, thePolycom 300 ($215 per unit). All the basic phones

» MISSION: Indemnifies makers of reality-TV programs

» GOAL: HNF is ready to test SIP-enabledVoIP systems

» TOTAL EMPLOYEES: 180» NUMBER WORKING REMOTELY: 50; 20

full-time telecommuters, 30 road warriors» NETWORK INFRASTRUCTURE:

LAN-based, with Layer 2 and Layer 3QoS enabled; 100-Mbps connections todesktops, with 802.3af PoE support; anda gigabit backbone connecting the restof the network

» REQUIREMENTS: VoIP system mustsupport at least 220 third-party,Ethernet-attached SIP-enabled IPphones, each with two 100-Mbps ports.All phones must support:» 802.3af and 802.3q/p and either

DiffServ- or TOS-based QOS» Unified messaging, ACD (automatic

call distributor) and presence» Telecommuters/road warriors

RFP SCENARIO: HAVENOFEAR INSURANCE, TAKE II

RReality TV’s popularity shows no sign of waning. That may be badnews for America’s collective IQ, but it’s a bonanza forHaveNoFear Insurance LLC, which indemnifies the sponsors, pro-ducers and staffs of these shows against lawsuits by traumatizedon-air participants. The company is still growing, but new com-petitors are coming online constantly. HNF finally has enoughmoney to purchase a SIP-compatible voice over IP system that ithopes will result in a competitive advantage.

Our updated HNF RFP calls for a SIP-capable IP PBX to handle220 Ethernet-attached phones, 50 telecommuters, 10 analog faxlines, 48 inbound digital trunks with DID (direct inward dialing)support, 35 outbound digital trunks and an ACD (automatic calldistribution) system. Finally, HNF wants to be able to use anythird-party SIP phone; we asked for recommendations on basicand executive phones, as well as attendant consoles and confer-encing phones.

Each vendor had to accommodate 20 telecommuters workingfull time from home over broadband connections and 30 road war-riors who use office phones but also travel extensively. We speci-fied presence management, conferencing and voice-mail capabili-ties. Finally, we asked about each vendor’s plan for applicationintegration and gathered pricing information.

Check out our full RFP and complete vendor responses atID# 1605rd2.

(VIT

AL S

TA

TS)

How many SIP phones does your organization have?

More than 5,0001,001 to 5,000201 to 1,00026 to 20025 or fewer

READER POLL


9%8%

21%38%

24%

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supported the G.711 and G.729 audio codecs.We found these IP PBX systems much easier to man-

age than their predecessors. Each vendor supplied a Win-dows 32-bit application to manage all aspects of call pro-cessing, users, phones and more. Each came with adatabase to store all information about users and devices,as well as facilities to view a graphical topology of thesystem and run standard and configurable reports. How-ever, none of the systems could monitor the health ofthird-party SIP endpoints, and none supplied us withreal-time information on network health or call quality.

Avaya, with its long history in telecommunications,delivered the most mature and feature-rich interface tomanage its S8300, the Avaya Site Administrator. Unlikethe other vendors, Avaya provided a rich CLI (com-mand-line interface) in addition to a GUI. The com-bined CLI-GUI was more difficult to master than theGUIs supplied by Zultys and Vonexus, but it gave us anextensive administrative toolset.

After examining the vendors’ RFPs and performingour hands-on testing, we gave Avaya’s SIP TelephonySolution our Editor’s Choice award. It provided thebest interoperability with third-party phones and the

best features-price balance. Zultys came in second; itearned only a fair interoperability score but almostmade up the difference by supplying the best softphone (MXIE) and the best price in our roundup.Vonexus came in just behind Zultys with a strongshowing in presence management and UM features. Itsproduct will make management and administrationeasy for any Windows shop, but it came up short oninteroperability and long on price.

Avaya IP Telephony Solution Avaya’s IP Telephony

Solution combines a G700 Media Gateway, an S8300Media Server and an S8500 Converged Com-munications Server (CCS). The company’sCommunications Manager (CM) provides call

processing and control from the media server to sup-port analog, digital, H.323 and SIP endpoints throughthe media gateways and servers. Support for multipleendpoints gave HNF the option to keep some analoghandsets until prices for SIP phones drop, later thisyear, we hope.

The CCS lends SIP support to the media server andgateway, and extends ACD (automatic call distribu-

B+

Marketcapitalization

Company name Year as of Feb. 9(stock symbol) founded Product name $000 Other Products

AVAYA (AV) 2000 Avaya Communication Manager $5,970,000 Avaya Meeting Exchange Audio Avaya S8300 Media Server and Web Conferencing Solutions Avaya G700 Media Gateway

Avaya Converged Communications ServerAvaya S8500 Media Server

Avaya Intuity Audix LXAvaya Communication Center Speech Access

Avaya IP SoftphoneAvaya 4602SW IP Phone

Avaya SoftconsoleAvaya SG200 Security Gateway

Avaya VPNremote client

VONEXUS, 2004 Enterprise Interaction Center (EIC) 67,410 Parent company offers contact A WHOLLY OWNED center automation, unified SUBSIDIARY communications/messaging and INTERACTIVE customer self-service solutionsINTELLIGENCE (ININ)

YearCompany name founded Product name Employees Other products Key customers

ZULTYS 2001 MX250 Enterprise Media Exchange 150 MX1200 Enterprise Media Exchange, MX25 Amazon.com, American Express,TECHNOLOGIES ZIP 4x5 IP Phone Modular SIP Gateway, ZIP 4x4 IP Phone, Coldwell Banker Elite, Conran

ZIP 2 IP Phone, BPS12 Battery Power Restaurants, Indecomm, Isuzu Supply, EPS48 Ethernet Power Switch, GM, Potlatch Corp., Thai AirlinesMXgroup, MXcluster, EPA3 Ethernet Power Adapter, LIPZ4 Linux SIP Soft Phone

PUBLIC COMPANIES

PRIVATE COMPANIES

VOICE OVER IP VENDORS AT A GLANCE

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tion), voicemail, IM and UM features to SIP end-points. It also can serve SIP trunks to a carrier serviceto take advantage of future SIP-based call origination

and termination. The CM sup-ports up to seven-digit dialplans, and Avaya says the sys-tem can expand with addition-al gateways and communica-tion servers, while retaining

centralized user and phone administration to reduceoperating costs.

Avaya proposed an SG 200 Gateway with VPN-remote Client software to connect HNF’s home users tothe central office securely via two-factor user authenti-cation and 3DES encryption. The SG 200 Gateway sup-ports stateful, multilayer packet inspection and protectsagainst DoS (denial of service) attacks. It also includes abuilt-in bandwidth manager for VoIP streams and sup-ports encrypted SIP signaling. Like Zultys and Vonexus,Avaya recommended the use of its PC application,Avaya’s IP Softphone R5, to supply IM and presencemanagement and facilitate call management.

IM and presence management are enabled by the

CCS and the IP Softphone R5 application. As with theother setups, the user l icense for most phonesincludes a soft-phone application. We could configureAvaya’s soft phone for real-time voice communica-tions using SIP, H.323 or even an analog line. UnlikeZultys or Vonexus, Avaya supports secure IM; the serv-er also provides IM logging and archiving to complywith industry regulations.

Avaya’s UM functionality came close to the full-fea-tured product supplied by Vonexus and easily surpassedZultys’ UM implementation. But unlike rivals, Avaya’sUM integrates with Exchange and IBM Lotus Dominoso users can employ their contact lists to initiate callsand chats. We could create, send and listen to e-mailmessages as well as access contacts, calendars and taskswith a TUI (Telephone User Interface) via text-to-speechfunctionality. And Avaya, like our other participants,includes a “follow-me, find-me” feature based on user-defined rules.

We especially liked Avaya’s “Extension to Cellular”option for mobile employees who primarily use cellphones. Users can map their desktop phone and cellphone together as a single extension, which means allincoming calls ring both phones and outgoing callsappear as if they’re initiated by the desk set.

In our hands-on portion of the testing, we man-aged the S8300 Media Server using the Win-32 AvayaSite Administration (ASA) application. ASA providedboth a GUI and a CLI to configure the network, callgroups, stations and system settings, including date,time, dial plan and class of service. We viewed, ranand scheduled all tasks from a directory tree; populartasks are also grouped in a favorites bar. We liked thewindow that displayed a history of our commandswith trace information about their execution on theserver or on endpoints.

Like the other vendors’ management apps, ASA gaveus handy wizards for common tasks like adding, chang-ing and removing users and stations. Although ASA letus manage all phones and their services, the interfaceunfortunately didn’t distinguish phone models andtheir feature sets—we had to make sure we knew whichstation and feature options were available on our testphones and configure them manually. But this was truefor Zultys’ and Vonexus’ offerings as well—automaticdiscovery of third-party SIP endpoints and their featuresets was not available from any system tested.

We managed the CCS from two separate Web brow-ser pages, one for administering users and resources,one for server maintenance. The SIP Managementpages let us import and export users, manage exten-sions, start and stop services, and configure settingsfor domain access and services like IM log settings.The Maintenance Web pages deal with hardware con-figuration, diagnostics and upgrades as well as databackup-and-restore and security. This bifurcatedarrangement enabled distributed administration. TheCCS also supports a firewall and a Tripwire tool that

Editor’sChoice

ComputingNetwork

��

Key customers News

Marion County Government, Gross revenue reported at $1.1 billion in fiscal Padcom Q1, up 18 percent from Q1 2004. Acquisition

of Spectel and Tenovis resulted in a loss of 13 cents per share in fiscal Q1.

Access Technologies, Parent company ININ reported $55 million in InterDyn Knowledgelake gross revenue for 2004, up 7 percent over 2003.

ININ is just barely profitable, reporting earningsof 3 cents per share in fiscal Q4.

Source: Yahoo.com, company reports

News

Company recently established distributorships in China and Australia. Newly established SIP Phone Center will distribute Zultys equipment in Europe.

Source: Company reports

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monitors and audits files on the server to verify theyhave not been modified or deleted.

We could integrate multiple G700 Media Gatewaysinto single logical service for redundancy. If onefailed, traffic was rerouted over surviving links auto-matically using Spanning Tree or port-redundancystrategies. A duplicate S8300 Media Server ($2,000)also can provide redundant call control. Finally, for$6,200 the CCS may be deployed in a duplicate serverconfiguration where each server shares the same IP

address to present a uniform interface to SIP clientswhen a failure occurs.

Avaya SIP Telephony. Avaya, (800) GO-AVAYA.www.avaya.com

Zultys Technologies MX250 Enterprise MediaExchange Of the third-party phones Zultys submit-

ted, only the executive model, the Polycom

SoundPoint IP 500, met HNF’s require-

ments. Its basic phone, the SNOM 105, does

WHOW WE TESTED SIP-COMPLIANT IP PBXSWe issued an RFP seeking SIP-enabled IP PBXs and required thatvendors set up their systems in ourSyracuse University Real-WorldLabs®. We asked them to bring threeSIP (Session Initiation Protocol)phones from three third-party ven-dors as well as an additional third-party endpoint, such as a videocon-ferencing unit or soft phone.Unfortunately, no one was adventur-ous enough to try videoconferencing.Avaya and Vonexus brought Xten’sX-Pro soft phone, while Zultys Tech-nologies went with Microsoft Mes-senger 4.7. Note that later versionsof Messenger no longer allow theoptional Communications ServiceAccount for SIP communication.

In addition to the endpoints, thevendors brought an array of phones

from Cisco Systems, GrandstreamNetworks, Polycom and SNOM Tech-nology AG (see “SIP Phone Choic-es,” page 49). Avaya played by therules and used Cisco, Grandstreamand Polycom phones. It also showedoff its own Avaya Model 4602SW,but since it’s not a third-party SIPphone, the 4602SW wasn’t used fortesting. Vonexus and Zultys did notante up the right number of phones:Vonexus used Cisco, Polycom andan analog phone with an Audio-Codes FXS gateway. Zultys stayedwith Polycom and SNOM.

After the vendors set up theirsystems, we ran a battery of tests torate the PBXs’ ability to interoperatewith third-party SIP phones (see“Test Battery,” below). We evaluatedeach phone for call initiation, termi-

nation, forwarding, conferencing andmore. We also scrutinized presencecapabilities and interrogated main-tenance features for MAC (moves,adds and changes), user and phonemanagement. After knocking our-selves out in the lab, we comparedthe RFPs (see our RFP and full ven-dor responses at ID# 1605rd2).

All NETWORK COMPUTING productreviews are conducted by current orformer IT professionals in our Real-World Labs® or partner labs, accord-ing to our own test criteria. Vendorinvolvement is limited to assistancein configuration and troubleshoot-ing. NETWORK COMPUTING schedulesreviews based solely on our editorialjudgment of reader needs, and weconduct tests and publish resultswithout vendor influence.

B_

Feature verification Action

Call initiation Call each phone

Message waiting indicator (MWI) on Call each phone, leave message, observe MWI on

Check voicemail and MWI off Check VM from each phone, observe MWI off

Conference call Engage all phones in a conference call

Call termination Call phone, let ring, hang up

Park and pickup Call park with one phone, pick up with another

Conference add One device adds another into conference

Auto camp One device camps on another

Dial busy Acknowledge busy signal

Call waiting Call device/receiving device puts call on hold, answers second call, returns to first call

Group call pickup Configure devices in a group and pickup incoming call from any member

Blind call transfer Blind transfer of incoming call

Managed transfer Manage transfer of incoming call

Call invalid number Acknowledge response

Transfer to invalid number Transfer call to invalid number

Conference invalid number Add an invalid number to a conference call

Call hold Listen to music on hold

Release call hold Release call on hold

Test Battery

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Avaya Vonexus Zultys Technologies

SIP Phone 1 Cisco IP Phone 7960 Cisco IP Phone 7960 Polycom SoundPoint IP 600

SIP Phone 2 Grandstream BudgeTone 100 AudioCodes MP-108A/FXS/AC/SIP SNOM 220*8-port gateway for analog phones

SIP Phone 3 SNOM 200 Polycom SoundPoint IP 500 Microsoft Windows Messenger 4.7

Soft phone Xten X-Pro Softphone Xten X-Pro Softphone

*Zultys also brought SNOM 105, 190.

IP Phone Choices

sport two 100-Mbps ports, but it lacks support for IEEE802.3af and 802.1Q/p. Maybe Zultys thought theSNOM 105’s low price—$200 with software license—could make up for these shortcomings.

Zultys’ submission for an attendant phone, theSNOM 220, fulfilled only our port and power require-ments; the 220 does not support IEEE 802.1Q/p, andin tests we couldn’t select an outgoing trunk to send acall nor choose an incoming trunk to answer or redi-rect a call.

On the plus side for Zultys is its MXIE, the bestPC-based application we tested. MXIE let us addattendant console functions to any SIP phone, and itcan handle as many as 16 simultaneous calls, bothinternal and external, including calls rerouted fromhome telecommuters. When we logged in as an oper-ator, the interface listed all users in our test “enter-prise” in a graphical display with presence manage-ment indicating who is on the phone, out to lunch orfree to chat. We transferred calls simply by draggingand dropping call appearances on displayed uses.Alternatively, MXIE provides 10-key access to call-ses-sion controls, such as transfer, hold and disconnect,so operators can manage calls without a mouse. AndMXIE is not only for operators. Zultys’ app, likeAvaya’s Softphone, is included with the softwarelicense for each SIP phone.

MXIE also provided presence management and IMfunctionality. Presence management and caller statuscan be automatic or manually driven, based on trig-gers, such as keyboard or phone activity. We couldinitiate and terminate calls, set call-handler rules,view the presence of other users, send instant mes-sages, and access voicemail and faxes from our desk-top. We could even configure MXIE to simultaneous-ly ring multiple phones.

Zultys’ telecommuting solution was clean and sim-ple. Using the ZIP 4x5 phone in home offices, HNF’stelecommuters establish IPsec tunnels to the centraloffice’s MX250. A VPN concentrator also may beinserted into the equation, but that was not includedin Zultys’ proposal. At home, the ZIP 4x5 phone pro-vides the functionality of a router with a firewall,DHCP and NAT support. Home users can connect theZIP 4x5 directly to a cable or DSL modem and set upa home network around it. The phone includes ananalog port in case of E911 emergencies and supportfor a Bluetooth wireless headset and voice-activateddialing. With this design, HNF may set up even morehome workers.

The MX250’s Linux-based OS performed well. Thedevice uses mirrored SCSI disks, and we could configureit with a standby unit. It also can support an N+1 clus-ter configuration of as many as four MX250s. The sin-

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Avaya Vonexus Zultys TechnologiesSIP Telephony Enterprise Interaction Center MX250

PBX featuresAdd-on conference Y Y N

Authorization codes Y Y Y

Automatic alternate routing Y Y Y

Automatic callback Y Y Y

Automatic camp-on N Y N

Automatic failover PSTN Y Y Y

Call waiting Y Y Y

Class of restriction/service Y/Y Y/Y N/N

Directed call pickup Y Y Y

E911 support Y Y N

Group call pickup/group paging Y/Y Y/Y N/Y

Paging Y Y Y

Trunk callback queuing Y Y N

Uniform dial plan Y Y Y

PresenceUser status/status updates Y/Y Y/Y Y/Y

Chat with user Y Y Y

Voice call/e-mail to user Y/Y Y/Y Y/Y

Camp on user Y Y N

Customizable status messages N N N

Y=Yes, N=No

SIP-Compliant IP PBX Features

gle MX250 system proposed to HNF scales up to 500users with no additional hardware necessary.

MX250 Enterprise Media Exchange, ZIP 4x5 IP Phone.Zultys Technologies, (408) 328-0450. www.zultys.com

Vonexus Enterprise Interaction Center 2.3Vonexus, like Zultys, built its proposal from the ground

up around open standards and SIP. The sys-tem is as feature-rich as our Editor’s Choice,but it costs more and lagged in our interoper-

ability tests. In addition, Vonexus didn’t bring thethird-party SIP phones we’d asked for, and the systemslacked HMP support, which means some features, suchas conferencing, may not work.

Vonexus was the only submission to use Windows asthe host OS in its IP PBX, the EIC (Enterprise InteractionCenter). This makes it easy for most enterprises to man-age. EIC enables NTLM security, provides an easy pathfor application integration and uses WMI for eventreporting and alerting. But don’t forget the care and

B_

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feeding required to block viruses and malicious code—imagine if not only your data-processing systems butalso your voice systems were possible attack vectors.

Vonexus advised HNF to use the Polycom 300 as abasic phone. This was the best basic model in ourroundup and had the lowest cost. Unlike Zultys’ basicentry, it satisfied our minimum criteria for power, portsand QoS (quality of service). It also had hard keys tiedto transfer, forward and hold calls with a dedicated vol-ume-control button. Vonexus, like Zultys, used Poly-com’s IP 500 phones for executives. Avaya maintainedboth courses (basic and executive) with the SNOM 200.It appeared Avaya was stacking the deck for its real pref-erence: the Avaya 4602SW.

Like Zultys, Vonexus added its user interface applica-tion, dubbed Interactive Client, to its attendant consolechoice, the Polycom IP 500. Even though the IP 500has a venerable set of features, IC ratchets it up a fewnotches. For example, the phone is limited to three lineappearances. (Line appearances are simply call appear-ances. For instance, while talking to one person, anoth-er calls. If you place the current call on hold and takethe next call, you now have two call appearances onyour phone set.) But with IC, line appearances are lim-ited only by the call resources on the server. And, wecould preserve server resources by limiting callresources on a per-user basis using a class-of-serviceconfiguration. Also, even though the phone does notsupport any bridged appearances, IC will give it a viewof 10 or more bridges.

Vonexus’ telecommuter offering was streamlined—theonly item required was IC, which Vonexus included in itspricing for the executive but not basic phones. IC costs$45 per user as an add-on to basic phones; factor in thatcost for the added functionality. Home users can use theirhome phones, cell phones or an IP phone with IC. With

the EIC’s “remote agent” functionality, we could route ICclient connections through a secure VPN, and associatedphones appeared as a direct extension off the EIC.

Vonexus’ presence management let us camp on some-one’s busy line; in other words, when the line becamefree, we automatically connected to it without having toredial. In addition, we could update presence using multi-ple methods, such as changing status via IC or a TUI.Users with the appropriate rights can update the status ofothers as well. And, unique to this roundup, presencemanagement is integrated with SMS (Simple MessagingSystem), which shows user status on Web pages. Vonexussays it plans to release a feature pack later this year thatwill enable integration with groupware and calendaring.

Vonexus gave us two redundancy options. If onlybasic call routing is required when a system is down, HNFcan use a backup proxy server from Vonexus that loadbalances incoming calls on a per-DID basis and routesthem to the proper locations. Maximum redundancy isachieved by deploying an automatic switch-over systemin the event of a planned or unplanned outage.

Enterprise Interaction Center. Vonexus, (888) 817-5904. www.vonexus.com NWC

Avaya Vonexus Zultys TechnologiesSIP Telephony Enterprise Interaction Center MX250

Unified messaging

Read voicemail from e-mail client Y Y Y

Caller ID in e-mail header Y Y Y

Delete voicemail from e-mail client N Y N

Listen to e-mail messages from phone Y Y N

Delete/forward e-mail from phone Y/Y Y/Y N/N

Follow-me, find-me Y Y Y

Microsoft Exchange support Y Y N

IBM Lotus Domino support Y Y N

Novell GroupWise support N Y N

Management

Graphical topology Y Y Y

Endpoint/network health monitoring N/N N/N N/N

Real-time voice-quality measurements N N N

Price $131,047 $176,345 $125,150

With installation and one-year maintenance $174,181 $194,631 Doesn’t directly supplyinstallation or maintenance

Y=Yes, N=No

SIP-Compliant IP PBX Features (continued)

SEAN DOHERTY is a technology editor and lawyerbased at our Syracuse University Real-WorldLabs®. A former project manager and IT engineerat Syracuse University, he helped developcentrally supported applications and storagesystems. Write to him at [email protected].

PETER MORRISSEY is a full-time faculty memberof Syracuse University’s School of InformationStudies, and a contributing editor and columnistfor NETWORK COMPUTING. Write to him [email protected]. Post a comment or questionon this story at www.nwc.com/go/ask.html.

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Network Computing, July 8, 2004

continues >>

Voice over IP is here to stay, as we reported inlast issue’s cover story (see “Holy EfficiencyEdge!” June 24, page 42, ID# 1512f1). But if you

want to run VoIP on your LAN or WAN, chances areyou’ll need to make some changes to your network,perhaps even redesign it. Although this may sounddaunting, keep in mind that VoIP is just an application,albeit one with some unique requirements.

Your network design and capacity can make or breakVoIP. Delay, jitter, packet loss and unreliability poseproblems for the real-time application (see “The VoIPBig Three,” page 73), so before you install your first VoIP

switch or phone, assess your network thoroughly. Anddo this several months before your VoIP rollout, just incase your network proves to need a major overhaul.

Your assessment should ascertain whether your net-work supports Layer 2 and Layer 3 QoS (Quality of Ser-vice) and VLANs (virtual LANs); measure your capacity;and determine if your network can support the addedbandwidth VoIP demands.

It’s especially important to make sure the networkcan handle extra voice traffic during busy hours. At aminimum, look at a full month’s data traffic to gaugecyclical changes in network activity by all the groups in

Before you install your first IP phone, make sure yournetwork is primed to handle it. Some key design and connection changes prior to installation can result in a smooth-talking VoIP system BY PETER MORRISSEY

Prepare Your Network

For VoIP

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your organization. Drill down into the data regardingpeak hours, and collect smaller samples of data thatshow one-minute or even five- or 10-second peaks.

Also look for peak calling times in your PBX’s call-detail records. You should be able to determine the num-ber of simultaneous calls you must support and howmuch bandwidth the VoIP application will add. Make surethere’s room for your existing applications and anticipatedgrowth as well. For help in calculating bandwidth for VoIP,see “Codecs: The Bandwidth + Delay Equation,” page 74.

Your switches and routers may have the necessaryVoIP features, but you still must inventory software codeversions and check that you have the most recentiterations. Expect to upgrade some software, because notall equipment comes VoIP-ready with bug-free versions ofQoS. But don’t get too hung up on QoS—think of it as aninsurance policy should you exceed the bandwidth.

Many VoIP vendors offer preinstallation assessmentservices and will recommend changes. This kind of helpwill save you some work and minimize finger-pointingif problems occur later on.

Reassess your network regularly, too. New applica-tions are added, people and departments move aroundand the network gets reconfigured, all of which canchange traffic patterns. Monitoring your network’scapacity and performance helps you avoid problems. Inaddition, VoIP monitoring tools can assist you inreassessing how well your network supports VoIP (see“VoIP Monitoring Tools,” page 74).

Back to BasicsMost VoIP vendors say duplexity mismatches—fullduplex on one end of an Ethernet connection and halfduplex on the other—are the biggest cause of VoIP per-formance problems. Check the duplexity settings ofyour connections, and use your network-managementsystems to check switch and router settings.

Start with the backbone connections, because these

have a huge impact on VoIP performance. Manually setboth ends of every backbone connection to full duplex.If you have both ends set to autonegotiate, you can getclues about duplexity from the error counters. If oneend experiences collisions, for example, it probably hasbeen set for half duplex. If the opposite end has lots ofCRC (cyclic redundancy check) errors and no collisions,it likely has been set for full duplex. A quick look at theconfigurations of most devices, especially switches androuters, will tell you their duplexity settings.

It’s best to test PC-to-switch and phone-to-switchconnections using autonegotiate if you have standard-ized on NICs and wiring-closet switches. If autonegoti-ate gets the highest speed available with full duplex,use it. If you want VoIP phones with built-in switches,choose ones that let you change duplexity settings.Also, check the connections from the phone’s built-inswitch to the PC and the edge switch.

To ensure that voice packets get priority treatmentand experience minimal packet loss and delay, turn onLayer 2 (802.1p) and Layer 3 (DiffServ) QoS throughoutyour network. Your VoIP equipment, including phonesand media gateways, should support QoS. Keep inmind that Layer 2 QoS settings are lost when the routerrebuilds the frame. Most routers can translate theappropriate Layer 2 QoS information using Layer 3 QoSbefore they transmit a packet—make sure your routerdoes this at wire speed.

If you’re plugging a desktop into your VoIP phone,be sure your PC doesn’t overwhelm your connectionwith data. This won’t be a problem if the phone’sbuilt-in switch and the wiring-closet switch support802.1p. If they don’t, a data transfer may interruptyour voice conversation.

Bottom line: Your QoS policy should give all VoIPtraffic first priority. The only exceptions are routingupdates and other control packets.

Avoiding FailureThe legacy phone network is well-known for its five-nines(99.999 percent) availability. A VoIP application, though,will be only as reliable as the data network supporting it.

Executive Summary

SVoIPSo you’re ready for the increased productivity andcost savings of voice over IP. But to reap the great-est benefit, you can’t just drop it in and start talking. IP voice needs the right network foundation so yourvoice calls don’t suffer from the deadly sins of VoIP—delay, jitter, packet loss and overall unreliability.Before you install the first IP phone, make sure yournetwork devices support Layer 2 and Layer 3 QoS(Quality of Service) and virtual LANs, and that yourexisting bandwidth can handle voice traffic. You’llprobably need to upgrade your device software, andperhaps your hardware, too.

Check all your network connections to ensureyou don’t have any mismatches of full duplex on oneend and half duplex on the other end of an Ethernetconnection. And remember that redundancy, both ofcore devices and power, is crucial to a smooth IPvoice implementation. Securing voice means puttingyour VoIP phones on their own VLAN using privateaddresses, plus minimizing and monitoring access toyour internal network.

And even after your VoIP system is up and running,keeping updated documentation of your data networkand VoIP components will help ensure IP voice quality.

QoS policy should give all VoIP traffic first priority, with the exceptionof routing updates.

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You can increase your data network’s reliability withredundancy, mainly at the network core, where itaffects the largest number of users. It also helps to addredundancy where your VoIP servers are connected. Ata minimum, you should have dual power supplies, dualCPU cards and mirrored drives at both the core and theedge of your network.

You should also connect redundant core routers toyour building switches with redundant connections(read how various vendors add redundancy, atwww.nwc.com/1209/1209colmorrissey.html). And to avoiddowntime, make sure you have spares for all criticalcomponents. With redundant equipment set up inhigh-availability mode, spares are put into productionwithin seconds (whereas it might take you hours toinstall new equipment).

Test high-availability schemes in your lab beforeyour rollout. In some cases, a backup CPU card in a

core router will need several minutes to reboot andload routing tables after the primary card fails. Somerouters handle this process more quickly.

Of course, adding redundancy adds cost. If you’rehaving trouble getting funding for this, help manage-ment understand the implications. Tell them howlong the network could be down when a replacementor repair is needed. Let the management team haveinput into determining how much downtime is acceptable.

Where possible, use failover standards such as theIETF’s VRRP (Virtual Router Redundancy Protocol) andthe IEEE’s 802.3ad link-aggregation protocol to connectredundant equipment to the edge. Interoperabilitywon’t be a problem with redundant connections link-ing two core routers, for example, because they’re fromthe same vendor. But standards are important at theconnection from the core to the edge, because you’re

TTo ensure your network voice isloud and clear, you need to knowthe three worst enemies of VoIP:delay, jitter and packet loss.

Delay is the time it takes fordata to get from Point A to Point Bon a network. VoIP is sensitive todelay because human conversationsoccur in real time; call qualitydegrades if a delay affects therhythm of the conversation.

The ITU G.114 recommends amaximum one-way delay of 150ms, or 300 ms round-trip. It’s best,however, to keep the round-tripdelay to no more than 250 msbecause at that point conversationquality declines. A VoIP phone typ-ically adds 60 ms of delay, but thisfigure varies by vendor and thesize of the jitter buffer (see“Codecs: The Bandwidth + DelayEquation,” page 74).

You can prevent the delay from overloaded router queues or an oversubscribed network bydesigning your architecture withgood QoS and enough bandwidth.Delay on slower WAN circuits andlong-distance circuits can be anissue, especially with overseasconnections. Although you canavoid some delay, you can’t eradi-cate it. Delay typically isn’t a LANissue (assuming the LAN wasdesigned properly), but a lower-

speed WAN circuit can add delay.Distance alone on a WAN circuit

will increase delay. The speed oflight adds at least 5 ms of delay forevery 1,000 miles of fiber. So acoast-to-coast connection of 3,000miles will have a round-trip delay of30 ms, which shouldn’t be a prob-lem. But the delay will be notice-able over international distances,VPNs or an overloaded network.

Delay also can magnify echo—aconcern even on conventional TDM(Time-Division Multiplexing) net-works. VoIP phones have built-inecho cancellation, but as delay isadded between the original speechand the caller, the echo becomesmore obvious. And echo-cancella-tion algorithms will have a hardertime filtering out echo. Ask yourVoIP vendor about its echo-cancel-lation features. Different vendorsuse different algorithms, and somework better than others.

Jitter is the variance in delay.It’s bad news for the same reasonas delay is: Voice is real time.When some voice packets arrivewith little delay followed by addi-tional voice packets with greaterdelay, parts of the conversation onthe receiving end will becomeuneven. Jitter buffers compensatesomewhat by buffering packetsand controlling how they are

played out, but if the jitter bufferis too large, it can generate toomuch delay. Codecs also may com-pensate by repeating the last por-tions of speech.

Packet loss can plague VoIP, too,but not as dramatically as youmight think. When voice is transmit-ted over an IP network, the voiceapplication typically uses the RTPprotocol running atop UDP (UserDatagram Protocol), which doesn’tretransmit lost packets. Retransmit-ting voice packets doesn’t makesense because voice is real time,and speech is broken up into suchsmall chunks that losing a few isbarely noticeable.

Some VoIP vendors say theirsystems can handle up to 10 per-cent packet loss. In general, it’sbest to keep packet loss in the 1percent to 2 percent range on theWAN and even less on your LAN. Ifyou rely on DTMF tones, you can’tafford any packet loss—one droppedpacket could cause a customer toget lost in an automated attendanttree. In that case, you’re better offusing TDM trunking on the portionsof the network that support anautomated attendant. Customersdialing into your phone system fromthe PSTN could easily be kept onTDM connections, since they arecoming in over TDM anyway.

THE VoIP BIG THREE

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bound to have equipment from different vendorsthroughout your network. A vendor-proprietaryfailover protocol could be best if the failover time isimproved, but weigh the trade-offs of using vendor-proprietary schemes versus standards.

You’ve Got the PowerVoIP requires backup power for your LAN equipment,

your IP phones and your VoIP servers. You probablyhave backup power in your data center. If so, it’s just amatter of upgrading the KVAs (kilo volt-amps).

Backup power is necessary in your wiring closets forphones and switches. You can power your phonesfrom the closet using the IEEE 802.3af Power over Eth-ernet standard, which uses Category 5 or better Ether-net wiring. Even a 15-minute backup will get youthrough a few brownouts or a short outage. Remember,though, that any outage will cause your switches andphones to reboot, possibly extending the outage for

several minutes and damaging equipment. It’s a goodidea to have battery-powered backup in the wiringclosets. (See “You’ve Got the Power,” at ID# 1508ws1,for more on PoE.)

Your VoIP system should be protected, especially fromInternet access—VoIP phones are computers and, there-fore, can be hacked. You can protect the phones by segre-gating them to their own VLAN using private addressesso that only the VLAN has access to the VoIP servers.

Obviously, you want to deny access to the phones

and servers from the Internet. Aside from the risk ofdenial-of-service attacks and possible break-ins,there’s also the danger of toll fraud—someone hijack-ing your phone and using it to make calls. You’ll alsowant to filter access within your internal network.Allow access to only those ports that are necessary,and make sure you have authentication and VPNs inplace for telecommuters who want to use the corpo-rate phone system from home.

Finally, to load IP configurations onto the VoIPphones, use DHCP. It also can be used to tell thephones where to download new software and config-urations via TFTP (Trivial FTP). If the phones rebootand cannot reach a DHCP server, they will go down,so the DHCP server’s reliability and availability areimportant.

It’s easy to get caught up in the technology, butprocesses are just as important. Before you make achange to your network—like adding a new feature orchanging a configuration—make sure you understandits impact on your VoIP system. Keep thorough,updated documentation regarding your data networkand the VoIP components so you can use that infor-mation for troubleshooting. Then your voice trafficwill flow smoothly over your network. NWC

VoIP Monitoring Tools» Agilent, www.agilent.com

» Brix, www.brixnet.com

» Empirix, www.empirix.com

» Shunra, www.shunra.com

» Spirent, www.spirentcom.com

TTwo different codecs are commonly used for VoIP.When planning your network, you need to know whichcodecs and related features are available from yourVoIP vendor. If you are only running VoIP on the LAN,G.711 (85 Kbps) is best because LAN bandwidth is soinexpensive. On the WAN, G.729 makes more sensebecause it uses less than half the bandwidth. Thecatch is that G.729 adds more delay. In the chartbelow, we’ve assumed a 20-ms packetization delay ateach end and a receive jitter bugger of 20 ms.

Some vendors implement VAD (Voice Activity Detec-tion) and silence suppression, which transmits packetsonly when users are talking. This reduces bandwidth uti-lization even further. However, speech may get clippedas VAD or the suppression switches on and off.

Beware that setting up voice calls generatesadditional traffic, as do control packets for RTP(Real-time Transport Protocol) that track the statis-tics. Integrating your VoIP with software applica-tions, too, adds traffic overhead, so test the applica-tions on your network. Make sure you have enoughbandwidth to support the maximum number ofsimultaneous calls that will occur during peak call-ing times, as well as the peak traffic utilization foryour existing applications.

Codec Total Ethernet Total Predictable Bandwidth End-to-End Delay

G.711 85 Kbps 61.5 ms

G.729 32 Kbps 75 ms

CODECS: THEBANDWIDTH +DELAY EQUATION

PETER MORRISSEY is a full-time facultymember of Syracuse University’s School ofInformation Studies, and a contributingeditor and columnist for NETWORK COMPUTING.Write to him at [email protected]. Post acomment or question on this story atwww.nwc.com/go/ask.html.

Protect your VoIP systemfrom Internet access—VoIP phones are computersand can be hacked.

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Network Magazine, August 1, 2004

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As IT assumes control of voice operations, networkarchitects must reconsider whether to transition to

VoIP and, more specifically, the Session Initiation Proto-col (SIP, RFC 3261). Even in cases where IT has investedin a new data infrastructure and now must replace end-of-life PBXs or build new sites, the decision to move toSIP-based communications is far from obvious.

While proponents have long argued for VoIP’s lowercapital and operational expenses, the case for a SIP-based telephony architecture has become evenmore compelling over the past 12 months.During that time, every major PBX vendor hasdelivered or announced a SIP migration plan.The IETF has also addressed showstoppinglimitations in the protocol’s basic voice capa-bilities, such as defining a clearer procedure for transfer-ring calls, and new RFCs and Internet drafts have addedmany of the boss/admin and attendant features longavailable in PBXs and vital to call centers.

On the other hand, there’s a big difference betweenwhat standards define and what products deliver. SIP isstill a relatively immature protocol, so network architectsshould expect the usual interoperability and deploymentchallenges. Although SIP spans some 100 RFCs and In-ternet drafts, most vendors will only implement a subsetof these, either because of a lack of resources or becauseof relevancy reasons. (For a list of who’s supportingwhat, see Resources on page 52.) Legacy PBX vendorsmay also extend SIP with proprietary protocols as ameans of protecting their existing markets.

As a result, SIP support means very different thingsto each vendor. Most implement certain “core” SIPfunctions, but the specific features and networkingconditions will vary between implementations. Havingdifferent vendors’ equipment run the same protocolsisn’t enough, as SIP messages may still contain slightsyntactical differences.

The outcome is interoperability problems that limitIT’s ability to second-source telephony equipment andtake advantage of competition’s major benefits—namely,lower prices and greater product differentiation.

Even when SIP servers and telephones do interoper-ate and implement the right standards, they may lackthe required telephony functionality. Least-cost routing,nonstop failover, reporting capabilities, and centralizedmanagement are just some of the critical features be-

yond SIP’s scope. Without these and other en-terprise-grade capabilities, even the most ro-bust SIP product portfolio is incomplete.

STANDARDS IMPACT Standardized VoIP telephony was supposed to

shake up the world of enterprise telephony, drivingdown equipment prices, increasing innovation in themarketplace, and giving IT the ability to turn to a sec-ond source for telephones and telephony equipment.

To some extent, SIP has achieved those goals. There area growing number of telephony clients available, includ-ing a wide range of telephones, as well as new softphonesand applications with embedded voice capabilities. Lookno further than Microsoft’s Live Communications Server(LCS) as an example of diversity encouraged through SIP.

Inexpensive telephony solutions are also available. En-terprise-grade SIP servers can be purchased today fromZultys Technologies and Interactive Intelligence, and forthe more adventuresome, a number of open-source SIPservers are available from such groups as Asterisk andSIPfoundry. With the open-source option, a fully func-tional SIP server can be outfitted for under $10,000 for 50users, including phones, auto attendant, and voice mail.

At that price, SIP servers could compare favorablywith today’s key systems used in the Time DivisionMultiplexing (TDM) world of small offices. However,SIP servers offer much more. They’re highly scalable by

Telecom vendors are supporting it. Enterprises are askingfor it. Is now the time to deploy VoIP’s hottest protocol?

SIP GOESPRIME TIME

BY DAVID

GREENFIELD

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nature, and since phones aren’t wired to a specific SIPserver, a failed server won’t necessarily crash the voicenetwork. Phones can locate and register with other SIPservers by querying DNS. Soon, a new Internet draftwill specify faster recovery from failures by defining aheartbeat message between an endnode—called a UserAgent (UA) in SIP parlance—and a SIP server.

SIP servers go beyond just voice. They can establishany type of session, laying the groundwork for deploy-ing a converged communications architecture capable ofdelivering voice, video, data, and IM. Not only does thisallow applications to blend multiplemodes of communications, but italso encourages a better user experi-ence within a given medium bycross-pollinating technologies. Thus,features commonly found in onemedium, such as call transfer in voice,can easily be extended to other mediums, such as IM.

MR. POPULAR Whereas many of SIP’s high-level benefits have been avail-able for several years, it’s only been in the past 12 monthsthat major PBX suppliers—namely, Nortel Networks,Avaya, Mitel Networks, NEC, Siemens, and Alcatel—havereleased their SIP strategies and products.

For the most part, SIP presents these vendors with areal pickle. Embracing SIP’s full functionality meansgiving away a tremendous competitive advantage, in-cluding the many years invested in proprietary callcontrol protocols, not to mention the lucrative businessthat comes from selling telephones and other end-points. At the same time, major customers of telephonyequipment, such as IBM and MCI, are pushing hard forSIP products.

The solution for most PBX vendors has been to intro-duce a SIP migration strategy hinged around a type ofSIP server called the Back-to-Back User Agent (B2BUA).A B2BUA, or what would be called a proxy server inthe non-SIP world, acts as two UAs, one mimicking thereceiver, and the other the sender. Some control logiclies in between the two. Within that control logic, aB2BUA can act on a SIP stream, converting it to thePSTN or a vendor’s own control protocol, for example.

By contrast, many SIP-only vendors implement SIPproxy servers (not to be confused with more generic

proxy servers). With SIP proxies, the SIP signalingflows directly between the two UAs.

There’s no question today that sophisticated SIP net-works will require some B2BUA functionality, if only toconnect to the PSTN. The broader question is whethernetwork architects will want to implement B2BUAs atthe core of their VoIP strategy.

In practice, the B2BUAs provided by PBX vendorstend to be more expensive to implement than SIPproxies, in part because of their architecture. SinceSIP proxies can step out of the signaling stream aftercall setup, they can maintain less state informationabout the session than the B2BUA. As such, SIP prox-ies require less hardware than a B2BUA and can scaleto support far more UAs given identical platforms.

What’s more, core functions needed for enterprisetelephony operations run on PBXs, not on B2BUAs,

Maturity

Feasibility*

Impact*

Risk

SIP is still in its early stages. Although much of the protocol in the core telephony area hasbeen specified, plenty of work still needs to be done for presence, IM, and moreobscure but significant cases of telephony.

Network architects can build a SIP-based telephony system today from productsprovided by individual vendors. If users require only basic call functions and the callsdon’t traverse NATs or firewalls, then multisourcing may be possible. However, IT wouldbe well-advised to run internal tests to ensure interoperability.

SIP stands to revolutionize communications architecture. Over the near term, the protocolhas encouraged the development of low-cost corporate telephony equipment. Newapplications such as Microsoft’s LCS are also incorporating presence structures based onSIP. Interoperability is still the key issue limiting SIP’s impact.

Deploying a VoIP telephony solution requires careful and deliberate planning. A datainfrastructure must be in place to deliver proper voice quality, ensure security, andprovide manageability. SIP-based networks are no different. The text-based nature of SIPplaces a bigger hurdle on product interoperability and will require enterprises to investin product testing to ensure a reliable network. Single-sourcing a SIP network helps, butisn’t sufficient alone.

* A S S U M E S L A R G E , D I S T R I B U T E D , S E L F - M A I N TA I N I N G C O M M E R C I A L E N T E R P R I S E W I T H N O S P E C I A L N E E D S R E L AT E D T O T H I S T E C H N O L O G Y

��

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RISK ASSESSMENT: SIP-BASED TELEPHONY

With TDM voice being so predictable,even seemingly rare hiccups can kill theSIP user experience.“ ”

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further raising system costs. So whereas Avaya’s Com-munications Manager implements the vendor’s least-cost routing algorithm, its SIP server doesn’t.

Security and reliability are other areas where SIPproxies may be better. Since B2BUAs rewrite thepacket’s innards, they represent a potential vulnerabil-ity. SIP proxies, on the other hand, only manipulate theinformation exposed in the SIP header. A B2BUA’s fail-ure is also more crippling than a SIP proxy, impactingall calls going through the box. With a SIP proxy, failureonly affects new calls. Existing calls or calls to deviceswhose addresses have been cached are unaffected.

Finally, SIP proxy servers are more adaptable thanB2BUAs. Since B2BUAs interrupt the media session, aUA’s features and functions won’t work if they aren’tsupported by the B2BUAs. This means that a vendor’stelephones may appear not to work with one another ifthey implement features differently than the B2BUA.Not so with a proxy, where the UA negotiates sessionfeatures directly.

GET PUMPEDAt the same time, extensive progress has been made inshoring up the SIP environment. Core telephony func-tions such as multiline or call waiting, call hold, calltransfer, and call conferencing are defined today, saysRohan Mahy, a network architect in Cisco Systems’Voice Technology Group and co-chairman of the IETF’sSIP Working Group.

Boss/admin functions such as simultaneously ring-ing multiple phone lines, barging into an active call,and whispering so that only one party can hear a super-visor or intercom are also standardized. The same goesfor attendant services such as auto attendant, attendantconsole, and night service.

Beyond the core PBX features, the SIP group has ad-dressed a number of other problems with the SIP envi-ronment. A number of RFCs and drafts deal with NATand firewall traversal, including RFCs 3581 and 3489.Authentication is clarified in RFCs 2617, 3261, and3310. Signaling protection prevents users from snoop-ing a packet stream.

REALITY HITSThere’s a big difference between SIP’s theory and itspractice. While SIP experts can manipulate the stan-dardized and drafted set of SIP methods and headers todeliver almost any call control feature, actual productimplementations lag behind.

Callback when not busy, for example, is possiblethrough an Internet draft that defines how a calling UAcan subscribe to the status of a recipient’s UA when therecipient is in the middle of a call. Once the recipientfinishes the call, the calling UA is notified and the twoare connected. Of the vendors contacted for this story,only 3Com claimed to support the draft.

With so many SIP features available, figuring out the

core functions of a SIP solution is critical. The SIPPING

16 has become a generic term used by some vendors

to reference those core functions. Exact definitions will

differ, but for Avaya the SIPPING 16 consists of:

• Call forwarding, both unconditionally and

when the phone is in a particular state, such as

being engaged in another call

• Call transfer, for both attended and unattended

calling using the SIP command, called the

REFER method

• Caller ID

• Call hold and music on hold

• Three-way and multiparty conferencing

• Call return, where you can return the last call

placed to your phone by dialing a sequence

or pressing a button

• Call park, where the phone call is put on hold

and picked up at another extension using the

NOTIFY command

• Follow-me, where calls are routed to a different

phone

• Find-me, where the system can try a series of

personalized numbers to locate the callee

• Call waiting

• An IVR system

• Multiple line presences

• Camp on, where you can subscribe to a

phone’s status and have your phone ring when

the targeted phone becomes available

• Call queuing, where calls can be placed on

hold and then served to the callee based on

some guideline such as first come, first serve

• ACD and redial

• Do not disturb

MCI’s SIP luminary, Henry Sinnreich, has co-authored

an Internet draft with a more exhaustive list of func-

tions (see Resources on page 52). More than PBX

features, he thinks phones should support:

• A Web page for phone configuration and a

personal phone book

• A presence indication, such as what’s found on

an IM client

• An IM screen, such as the large color ones on

Windows CE 5.0 phones

• A means of entering URLs

The Phantom 16

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29 AVAYA PLAYBOOK


Part of the problem is that SIP addresses so manydifferent requirements. SIP specifications span thewired and wireless worlds and enterprise and carriersnetworks, not to mention a wide range of technologiessuch as voice, IM, and presence. Vendors often need asubset of these specifications to meet the needs of theirtargeted markets.

On top of that, SIP vendors must devote considerableresources toward tracking, contributing to, and imple-menting a given SIP standard. Factor in the myriad ofconfiguration possibilities, and it’s easy to see why ven-dors carefully select the RFCs and drafts they implement.

Call transfer, for instance, consists of two functions:basic or unattended transfer, where a third party directsa caller phone to ring another phone; and attendedtransfer or consultation with hold, where the thirdparty places a caller on hold, contacts the callee, and, ifallowed, transfers the caller to the callee.

Take into account the different types of NATs, poten-tial users’ states, and other conditions, and there aredozens of potential call transfer cases that need to beconsidered here. For example, problems can arise whentwo callers are behind NATs, when individuals receivingtransferred calls forward them to voice mail, or when anattended transfer turns into an unattended one.

Call transfer is so fundamental that vendors mustsupport the operation, but it’s easy to see why vendorsaren’t rushing to implement every SIP function, or whythey might choose to write their own SIP extensions.PBX vendors such as Avaya and Siemens claim to sup-port SIPPING 16, a set of core SIP features (see “ThePhantom 16” on page 51), but they implement other fea-tures through proprietary extensions. Avaya adds atleast 30 other features to its mix, including last numberdialed, malicious call trace activation, priority calling,call parking, and call pickup.

That’s a far cry from Cisco. Although an early pio-neer in the protocol and still a strong proponent of SIP,Cisco lacks a significant SIP implementation within theenterprise. The company’s Cisco Call Manager (CCM)offers a SIP trunking interface that can connect to a SIPproxy server targeted at the carrier’s network, but thevendor doesn’t provide a SIP proxy server with the fea-tures required by enterprises.

INTEROPERABILITY Given that SIP is a text-based protocol and very flexible,interoperability remains a challenge. “We had a wholetelecom network crash just because Cisco’s SIP proxyimplemented an INVITE message with a nonstandardperiod in it,” says Baruch Sternman, CEO of KayoteNetworks, a new VoIP start-up specializing in deliver-ing SIP security and protocol correction services.

In other instances, features may be implemented indifferent ways. Call parking is a good example. Thereare two ways to park a call in SIP, and unless the park

server and telephones support the same approach, thefunction won’t work.

The SIP community is aggressively pursuing thesecompatibility issues, both formally and informally. Atthe end of last June, for example, Nortel announced itwas opening up the code for its SIP proxy to enable bet-ter interoperability with third-party clients. The com-pany says Polycom, Texas Instruments, cordless phonemanufacturer Uniden, and i3 Micro, a vendor of resi-dential IP gateways, will align their products with itsMultimedia Communications Server (MCS). As for theIETF, this month it will conduct its 15th SIP interoper-ability test in just over a year and a half.

Still, with TDM voice being so predictable, even seem-ingly rare hiccups such as the ones enumerated here cankill the SIP user experience. Deploying VoIP is riskyenough without raising the ire of a user or line-of-businessmanager. For those reasons, enterprise-class SIP systemsmust implement far more than just RFC 3261. Vendorsmust also support the more fluid Internet drafts, as well asadd functions outside of SIP’s scope, such as least-cost callrouting, reporting capabilities, and management functions.

In the end, if a single-sourced, SIP-based solution isthe goal, any number of options are available, but thewhole point of SIP is to move beyond single-sourcing.In truth, IT is probably less interested in building a net-work a priori from a gaggle of vendors’ gear than in pre-venting vendor lock-in.

Multisourcing is possible if user expectations are setappropriately. Instead of supporting every marginal fea-ture some of the time, network architects need to iden-tify which core features are required and ensure thatthey work all of the time. Even then, vendors shouldclearly state which of their products and version num-bers have been proven to be compatible and under whattypes of networking conditions.

E x e c u t i v e E d i t o r D a v i d G r e e n f i e l d c a n b e r e a c h e d a t

d g r e e n f i e l d @ c m p . c o m .

resources : : :For more information about SIP, check out The SIP Centerat http://SIPcenter.org/. It provides a wide range ofmaterials, including product lists and white papers.

Rohan Mahy’s PowerPoint presentation on the statusof SIP standards can be viewed at http://i.cmpnet.com/networkmagazine/docs/sipupdate.ppt.

A list of the RFC and Internet drafts supported by lead-ing IP PBX vendors can be found at www.networkmagazine.com. Just type in Doc ID #1908f4 into theArticle Locator field.

MCI’s Henry Sinnreich has co-authored an Internetdraft on which features to look for in SIP phones. Youcan find that draft, draft-sinnreich-sipdev-req-05.txt, atwww.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-03.txt.

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30 AVAYA PLAYBOOK

Network Magazine, December 1, 2004

continues >>

VoIP RACESFOR THE REMOTE OFFICE

Most of us believe VoIP can reduce the telephonycosts of remote branch offices. What’s harder to deter-mine is which kind of VoIP we’re talking about. Is thisthe VoIP that’s built around a remote gateway? TheVoIP that involves only IP phones in the branch of-fice? Or is it a combination of both? Makingan informed choice is difficult becausethere are so many ways VoIP can beimplemented.

To help sort through those branchoffice telephony options, we askedsix VoIP vendors to price out a 20-user branch office with a T1 con-nection to the main site for voiceand data, and two analog lines forlocal faxing and emergency calling.We collected their responses based onlist pricing, analyzed the results, and ex-tracted some hard bottom-line figures (seetable on page 63).

What we found was that no single vendor today of-fers the ideal branch office solution, though each ap-proach has some of the right elements. ShoreTel offers adistributed architecture that allows remote sites to con-tinue to function together even if a main office fails.Cisco Systems builds VoIP into routers, providing aneconomical package that’s tightly integrated with thedata infrastructure. Zultys Technologies’ Session Initia-tion Protocol (SIP)-based architecture allows enterprisesto use any SIP-compliant product. Avaya offers a uniqueapproach to ensuring high-quality voice that’s well-suited to gracefully transitioning existing environmentsto VoIP. Nortel Networks and Alcatel’s solutions alsofare well in this last respect.

Choosing the right architecture requires boning up onthe basic branch office VoIP architectures. The differenceshave less to do with specific protocols and more to dowith the amount—and cost—of equipment needed in the

remote office. The easiest and cheapest solution forrolling out VoIP to the branch is registering re-

mote phones or softphones with a telephonyserver in the main office. Fault tolerance

can be provided by pulling dual-accesslines to the remote premises. Whereredundant access lines aren’t neces-sary, too expensive, or just insuffi-cient, network architects can locate agateway in the branch.

Ultimately, emerging hosted serv-ices from the likes of MCI will help

address the troublesome problem of re-mote site telephony without losing tight

integration with the corporate VoIP architec-ture. When that happens, network architects will be

able to tie the hosted VoIP services at the remote sites tothe telephony services at the central office.

JUST THE PHONESOne of the best uses for VoIP is in supporting the small-est offices. A few phones can be located at any sitewithout additional CPE. The phones register with a te-lephony server in the main site over a DSL or cable In-ternet connection, much as is done with consumer VoIPservices such as Vonage.

Admittedly, a similar approach is possible in the TDMworld, but it’s expensive. Voice architects can tie remotephones back to the main PBX through Off Premises Exten-sion (OPX) circuits provided by the carriers. Those circuits

A SLEEK SET OF IP PHONE SYSTEMS AND SERVICES PROMISESTO BETTER SERVE REMOTE OFFICE WORKERS. F IND OUT WHICHIS RIGHT FOR YOU. | BY DAVID GREENFIELD

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continues >>

turn the remote phone into just another extension, but at aprice. OPX circuits from BellSouth, for example, run $72per month per location, plus $60.48 for the first mile be-tween the locations and an additional $3.89 for each sub-sequent mile. It’s little wonder that BellSouth now recom-mends users connect back over DSL.

By avoiding the cost of remote CPE, network archi-tects can shave their per-user costs by hundreds of dol-lars. Here, SIP-based architectures such as that fromZultys have a big advantage. Today, most major net-working vendors offer nominal SIP support, typicallywith proprietary extensions for advanced functions.The problem is that while third-party SIP phones maywork with the telephony server, they’ll either do so atreduced functionality, or require a server upgrade to getfull SIP functionality.

Not so with Zultys’ MX250. As an all-SIP-based so-lution, it lets network architects choose any SIP end-point. Its own ZIP 4x5 phones bundle in a firewall,router, NAT, VPN, and built-in Ethernet switch for$400. Third-party SIP phones can also be purchased atjust about any price point, depending on the feature setdesired. This can dramatically reduce the cost of a re-mote site installation. Instead of spending $300 for aproprietary IP phone, customers can purchase a SIP ver-sion for a third of that price.

Attendant consoles can drop those costs even further.Attendant consoles generally cost about a hundred dol-lars more than a regular phone. Avaya still charges $500for its software-based attendant console used with itskey system replacement, IP Office. However, many IPtelephony vendors now build this functionality into thephone, or, as is the case with ShoreTel, bundle it as anapplication with the phone.

The difference is particularly acute when it comes tohigh-end phones. Today’s third-party SIP-based soft-phones, such as Xten Networks’ EyeBeam, run around$60, equipped with integrated video, presence, and IMcapabilities. A comparable hard client would cost hun-dreds of dollars, and even Avaya’s SIP-based softphonecosts twice that amount.

“Telephony vendors make 35 percent of their rev-enue off the handset,” says a marketing manager at oneof the major telephony vendors. “A VoIP handset witha color screen will run you $600 to $900. Why wouldyou pay that for a phone? You could buy three Dell PCsfor that price.” Unfortunately, there’s no getting aroundthe software licenses with some phone systems. Half ofAvaya’s phone-related charges, for example, come fromsuch licenses.

Softphone adoption may sound unlikely at first. Tosome extent, this is part cultural and perhaps part gen-erational. The likelihood that calls will be missed or in-terrupted because of a system reboot is far lower nowgiven the relative stability of Windows XP. Today, typi-cal college graduates expect that sort of stability.They’re also used to communicating through IM andWebcams and using all the functionality of mobilephones. Using softphones comes naturally to them.Even many of today’s traveling workforce and call cen-ter agents use softphones for specific applications, sowhy not the general knowledge worker in the office? Nodoubt some users will balk at the idea and demand astandard handset, but over time that will be the excep-tion, not the rule.

Still, registering remote phones with a central te-lephony server has its limitations. There’s no way toconnect those phones to the local PSTN in the event of

Router

Ethernet switch

with inline

power

Gateway or

telephony

server

Endpoints

Solution total

Pricing Out the Remote Office : : :Cisco SystemsAlcatel Avaya ZultysNortel Networks ShoreTel

$7,690

$2,390

N/A

$4,900

$14,980

$1,800*

$2,400*

$2,353

$12,524

$19,077

$1,800*

$2,400

$5,800

$14,000

$24,000

$1,800*

$2,400*

$3,360

$8,400

$15,960

$1,800*

$3,495

$6,300

$7,400

$18,995

$1,800*

$2,400**

$1,995

$11,565

$17,760

* Added to bid for comparative purposes. All prices are list prices and for the branch office only. Pricing for equipment in the main

office isn’t included. Actual street prices will vary by 5 to 35 percent depending on the contract.

** ShoreTel pricing provided by Xtelesis, a ShoreTel reseller.

An expanded version of this chart is available online. Search for Doc ID# 1912f4 at www.networkmagazine.com.

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continues >>

a failure, so lose the DSL connection and telephoneservice is lost, too. Depending on customer require-ments, this might not be a significant problem. If remoteworkers have cell phones, the temporary loss of theirdesk phone may not be a big deal.

A bigger concern can be the lack of a local telephonenumber on centrally trunked remote office phones. Acall to the local pizza parlor must first be sent to the cen-tral telephony server, put onto the PSTN, then deliveredback to the local dialing area, possibly as a long distancecall. What’s more, E-911 calling may not function prop-erly. Carriers also have a problem delivering local phonenumbers across the WAN. If an office in New Jersey re-ceives a dial tone from the main office in San Francisco,New Jersey Direct Inward Dialing (DID) phone numberscan’t be provided through the San Francisco office’sphone system.

REMOTE SITE SURVIVABILITYAs branches grow in size and importance, IT must pro-vide a more fault-tolerant local telephony solution.One IT architect contacted for this story addressed theproblem by adding dual T1s to a remote office. Whilemore expensive than the capital costs of adding re-mote CPE, redundant access lines improve the avail-ability of data, as well as the voice infrastructure. It’sworth noting that dual-access lines don’t necessarilyincur twice the price of a single line. MCI’s InternetDedicated T1 Shadow service, for example, adds adormant T1 that’s only activated when the primary T1fails. (The Shadow T1 can also be diversely routed tothe site, further increasing reliability.) The service ispriced at a lower rate, typically another $525 permonth assuming a local loop charge of $240 and a portcharge of $285. There’s also a $50 installation charge.

In contrast, a regular T1 usually costs $970 per month,with a $700 installation fee.

Where local survivability is critical, most organiza-tions will opt for locating a gateway at the remote office.The gateway connects to the PSTN, the LAN, and, de-pending on the configuration, the incoming access line.The phones within the office register with the gateway,but all the call processing happens back at the teleph-ony server in the main office. In the event of a data net-work failure, the gateway redirects calls to the PSTN,thus providing backup calling functionality.

Of course, locating additional CPE boosts equipmentcosts and introduces remote site management issueswhile delivering different degrees of telephony func-tionality. The most expensive approach was taken byAvaya, Nortel, and Zultys. Avaya’s G350 Media Gate-way, for example, comes with the S8300 Media Server,a backup telephony sever that Avaya refers to as a LocalSurvivable Processor (LSP). The server prices in at$5,800, about the same as Cisco’s 2811 router with inte-grated voice capability. When the main line fails, theLSP functions as a mini-PBX, providing full telephonyfunctionality. Nortel takes a similar approach with itsSurvivable Remote Gateway (SRG), as does Zultys withits MX25. However, the MX25 can’t accept local voicemail or auto-attendant.

Using a gateway in the remote office can also provideimproved voice quality. Avaya’s G350, for example, canseparate the channel carrying data between the phones(called the bearer channel) from the signaling used to setup the telephone connection. The data portion movesacross the PSTN, while the signaling necessary for settingup phone calls or performing various telephony func-tions travels on the WAN. This split arrangement givesremote users on a poor DSL connection the voice quality

Maturity

Feasibility*

Impact*

Risk

Branch office VoIP technology has gone through the early trial, early adopter phase. Thephones are stable, and the gateways are deployable. However, many phone systems stilllose some functionality when operating in backup mode. In addition, presence technologyand integration with other media—the raison d’etre of VoIP—is still spotty.

While VoIP phones are out there, architecting a mixed voice-data network isn’t easy.Network architects need to invest the time to engineer the WAN and LAN correctly foroptimum voice quality and availability. Using third-party SIP products obviously complicatesinteroperability and deployment, but all the vendors today provide end-to-end solutions.

Branch office VoIP systems already provide significant benefits in some companies byhelping them cut operating costs and reduce capital expenditures in new offices. With acommon telephony system, customer calls to one office can be transferred to another. Userscan call from any phone as if they were calling from the one on their desk, and mobileworkers can gain the full functionality of their corporate phone system while on the road.

The risks of rolling out VoIP in a new branch office are fairly low. The bigger challenge comesin justifying and deploying VoIP corporate-wide, or in replacing an existing, functional PBX.Once a decision has been made, engineering the network is also a challenge. Perhaps thebiggest issue is getting buy-in from line-of-business managers. Involving them in thedecision process will go a long way toward managing expectations down the road.

* A S S U M E S L A R G E , D I S T R I B U T E D , S E L F - M A I N TA I N I N G C O M M E R C I A L E N T E R P R I S E W I T H N O S P E C I A L N E E D S R E L AT E D T O T H I S T E C H N O L O G Y

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RISK ASSESSMENT: VoIP FOR REMOTE OFFICES

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33 AVAYA PLAYBOOK


continues >>

of the PSTN, with access to all the telephony functions ofthe office phone system.

Cisco’s survivability solution takes a middle-of-the-road approach to pricing and functionality. Although itcosts about the same as Avaya, Nortel, and Zultys’ solu-tions, Cisco’s includes routing and Ethernet switching(worth about $4,200), as well as local voice mail. (Avayasays its product can perform remote routing, but few en-terprises rely on the company for routing.) When themain link fails, Cisco’s 2811 with Survivable RemoteSite Telephony (SRST) code provides the PBX function-ality, but only to a limited number of phones and with-out any of the policy and calling intelligence of the coresite. Nothing, for example, would prevent users from di-aling a 1-900 number when in survivable mode.

IT architects have long been reluctant to load remotesite routers with new applications because of the im-pact they may have on routing performance. As part ofCisco’s new line of routers announced last fall, the 2811can be equipped with enough additional Digital SignalProcessing (DIP) resources to run the codes needed toconvert between VoIP and analog data. With Cisco’sSRST technology activated in the router, the 2811 canact as a backup telephony server for the Cisco Call Man-ager (CCM) servers located in the main office.

With a single device containing the core remote of-fice connectivity, IT can cut shipping costs and installa-tion time, as well as simplify ongoing management. Atthe same time, however, there are administrative issuesto consider. For one thing, there’s “one neck to choke”if something goes wrong, notes a major Cisco customer.For another, there’s now a shared device between thevoice and data infrastructure. If you’ve outsourced yourremote voice and data maintenance to differentproviders, that common element will pose a problem.Finding an organization with credible experience man-aging both voice and data is possible, but not easy.

The next step down in pricing and functionality forremote site survivability falls to Alcatel. The Alcatel IPRemote Unit X-IP now includes an additional processor.The gateway at the main office calls all the branch sitesand re-establishes the signaling connection, providingfull telephony functionality. The problem is that an ana-log trunk is consumed in the process, which in our casewould leave the remote office with a single trunk forcalling. Alcatel is developing a gateway with a backupprocessor due to be released in 2005.

NETWORKED SERVERSWhen remote branch calls need to be localized andthere’s a number of branches to manage, some ven-dors, such as Avaya, can equip the branch with a low-end telephony server. All the call processing andvoice mail is kept locally at the branch, while admin-istration can still be centralized at the main office.Avaya’s IP Office 403 DS runs just over $12,000 for 20

When pitching VoIP for the branch office, justifying

the expense of the project can be just as important

as the implementation itself. So what’s the best way

to approach such as delicate subject? We posed that

question to several network architects who have

made the transition to VoIP.

One major theme that emerged was that while

VoIP cuts costs, its biggest benefit is in the functional-

ity it delivers. “At a comparable price point to a key

system, Avaya’s IP Office enabled us to give our

branch offices PBX functionality,” says Jack Nelson,

manager of network services and operations at ap-

pliance manufacturer Sunbeam Products. “We can

now provide auto-attendant and call routing, create

small ACD groups, and in general bring a lot more

flexibility to addressing individual requirements.”

Conferencing was another big savings for Sun-

beam. Nelson says the company spent up to $130,000

annually on AT&T’s conferencing services. By deploy-

ing Avaya’s IP Office at branch sites, users were given

the ability to set up ad hoc conferences at just

$40,000, resulting in a first-year savings of $90,000.

Similarly, adding unified messaging can double the

cost of a PBX or key system, whereas that feature is

often included in VoIP solutions.

Beyond the specific costs, lower maintenance is

another big motivator. With TDM sets, technicians are

needed to change the paper labels on phones or

physically move the handsets. An engineer at a bank-

ing-related company estimated that this added an-

other $100 per user move, not to mention the cost as-

sociated with fixing any additional problems created

by the technician. With VoIP, the phones automatically

reconfigure themselves once plugged into the net-

work, so users can move their own phones.

VoIP also allows companies to consolidate cer-

tain services. “VoIP has allowed us to centralize our

live receptionists into a call center across regions,

producing a net savings of $100,000 annually,” says

Ryan Colbert, director of IT at Rissman, Weisberg,

Barrett, Hurt, Donahue & McLain, a law firm based in

Orlando, FL. Instead of calls coming into each office,

calls are now answered out of one office and then

transferred as necessary either within the office or

across the Internet to the firm’s other offices. Col-

bert could have eliminated those positions using an

auto-attendant at each office, but that would have

meant missing the personal touch that the firm’s cus-

tomers require.

VoIP: The Business Case

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continues >>

Four Approaches to Remote Survivability

Mainoffice

WAN PSTN

Phone

X

Survivability In the GatewayA backup processor in the remote gateway provides

all telephony functions during an outage.

GatewayRouter

Switch

Mainoffice

WAN PSTN

Phone

X

Survivability In the RouterCisco relies on its new remote routers to provide Ethernet

switching and backup call-control, saving the cost of

an additional gateway.

Router and switch

Mainoffice

WAN

Phone

X

Survivability In the Main OfficeAlcatel avoids the high cost of a backup processor at the

remote site by processing the call signaling in the

main office’s gateway.

GatewayRouter

Switch

Mainoffice

WAN PSTN

Phone

X

Survivability In the Distributed PBXShoreTel uses a distributed architecture, where the telephony

servers at each site work together as a single system.

Telephonyserver

Router

Switch

Phone

PSTN

Phone

Phone Phone

Key: Media path Signaling path

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35 AVAYA PLAYBOOK


users, excluding any additional applications such asunified messaging.

Functionally, this approach is similar to having a setof autonomous PBXs networked together. The problemwith this implementation is that each IP Office must bemanaged separately, and functions such as paging be-tween phones won’t operate between offices. By con-trast, Avaya’s Communication Manager allows architectsto affect changes globally across the entire network,rather than having to manage each site discretely.

Over the long term, that’s likely to change. Avaya isworking on enabling functions such as call processingto be distributed across gateways, with the aim of givingarchitects the manageability of Communication Man-ager, combined with the distributed capabilities of IPOffice. Sources close to the company expect that func-tionality to be introduced some time next year.

Meanwhile, network architects will find a similar ap-proach available today from ShoreTel in the form ofShoreGear, a distributed PBX. With ShoreGear, call pro-cessing occurs locally at each site, but all sites are treatedas a whole for the purposes of management and adminis-tration. As a result, less traffic needs to travel back to thecentral telephony server, improving scalability and re-ducing the amount of WAN bandwidth consumed at theremote site. On the other hand, network architects willwant to be sure that the update traffic needed to keep the

ShoreTel servers current doesn’t outweigh those band-width savings. The ShoreTel solution is also among themidtier, coming in at $17,760 for 20 users when adjustedfor Ethernet switching and routing costs.

SERVICE AND INFRASTRUCTURE Alternatively, the problem of managing remote officeusers may be offloaded onto hosted service providers.The widespread deployment of SIP trunking will allowIT architects to tie central telephony servers into a serv-ice provider’s hosted voice service offering. IT could re-tain control of telephony operations and let serviceproviders manage the small offices that are too costly toserve in other ways.

By March of next year, MCI says it will provide suchintegration with its MCI Advantage service. Dependingon the existing IT environment, remote office manage-ment costs could be reduced significantly, while stillmaintaining a single corporate-wide telephony service.

MCI must keep in mind, however, that simply pro-viding the IP equivalent of OPX lines won’t be suffi-cient. MCI must also provide full presence and conver-gence integration, along with basic telephonysupport—a tall order indeed.

E x e c u t i v e E d i t o r D a v i d G r e e n f i e l d c a n b e r e a c h e d a td g r e e n f i e l d @ c m p . c o m .

November 5, 2004

By Paul Travis

Grant Thornton International puts hundreds of taxand accounting advisers on the road every day tomeet with business clients and provide guidanceon financial issues. To make it easier for thosetraveling workers to stay in touch with clients andco-workers, the firm deployed a unified-messagingsystem this week that will save hundreds of thou-sands of dollars and make it simple to send,retrieve, and work with a variety of message types.

The firm, with 50 U.S. locations and around 3,700employees, deployed Unified Messenger softwarefrom Avaya on four Compaq servers to provide itsadvisers with a single mailbox for all voice mail, E-mail, and fax messages. The system lets themretrieve voice messages on their PCs; have E-mails read to them over the phone using text-to-speech conversion or have E-mail and attach-ments sent to a fax machine for printing, andretrieval, editing; and send faxes while on the road."We used to have E-mail servers at 48 locationsand a lot of costs associated with servers, soft-ware, and administration," says Kevin Lopez,national telecommunications manager. "Now weonly need four servers to cover the entire nation,and we're using the bandwidth on our WAN tocarry the traffic."

The move is saving the firm $400,000 on the costof servers. Additional savings comes from reducedmaintenance and support. "It also provides ourpeople with more time during the day to work withclients," Lopez says. "I figure it will pay for itself ina year and a half."

The goal is to give mobile workers access toeverything they would have in the office by addingthe messaging software to their laptop computers."The next logical step will be to provide the mobileworkers with an IP phone that they can plug intotheir laptop," Lopez says, although he doesn'thave a timetable for that move.

Grant Thornton, which has 585 offices in 110countries, is moving cautiously toward unifiedcommunications and will maintain separate voiceand data networks for a while. "The easiest way tomaintain quality of service is to run two T-1s, onefor voice and one for data. It may not be the low-est-cost approach, but it provides better security,"Lopez says. He's worried that the security threatsand denial-of-service attacks that hit data net-works would harm the firm's ability to maintainvoice communications if he ran everything on onenetwork. "Why make big changes that make your-self more vulnerable?"

Tax Firm Deploys UnifiedMessaging

Grant Thorton looks to put traveling workers more in touchwith clients and co-workers.

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36 AVAYA PLAYBOOK

informationweek.com, November 5, 2004

August 24, 2004

An Avaya Internet Protocol telephony communi-cations solution lets the good folks at NBCOlympics stay in touch over a private, wide areanetwork that links the company's headquartersin Stamford, Conn., and the InternationalBroadcast Center and field offices in Athens.NBC staff and sponsors can dial colleagues,friends, and family in the U.S. without incurringinternational toll costs.

Under a contract that began with NBC's broad-casts of the 2000 Olympics in Sydney, Australiaand will include the network's broadcasts fromthe Beijing Games in 2008, Avaya is delivering itslatest communications technology for each NBCOlympics broadcast. This year that means trad-ing the broadcaster's traditional PBX communi-cations platform in for an Avaya Media Serverfeaturing a software-based IP telephony solution.

"We were able to configure and rack-mount theserver before shipping it to Greece, so when it

arrived all we had to do was roll it out and hookit up," said Bob Kiraly, Director, Broadcast andTelecommunications Operations for NBCOlympics. "That meant a significant time savingsand will make it simple to move the setup to anew location when the Games are over."

As the centerpiece of NBC Olympics's network,the Avaya Media Server feeds Avaya'sCommunication Manager software to gatewaysat each site, linking them over a single data net-work and eliminating the need for separate com-munications switches. Communication Manager,Avaya's flagship IP telephony software, is anopen, standards-based software platform thatsupports both IP telephony and traditional voice.

Avaya Global Services installed the network forNBC Olympics and provides remote monitoringand around-the-clock service and supportthrough a dedicated project manager and a teamof on-site technicians.

Avaya Shoots for Olympic GoldAn Avaya-powered IP network is handling the telephone needs for

NBC's broadcast of the 2004 Summer Olympics.

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37 AVAYA PLAYBOOK

commweb.com, August 24, 2004

cmp ip communications · $30,000 cost and disruption of recabling the central library by using our...

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