Sound Solutions: Defensible VoIP Options for Companies: Defensible VoIP Options for Companies
Articles and Tips:
01 Feb 1999
When asked, "How are network administrators responding to voice over IP (VoIP)?" NetWare Connection's sources provided conflicting viewpoints. For example, Jeff Tyre, product manager of Nortel Networks' V/IP Phone/Fax IP Gateway, claims that network administrators are "really enthused" about VoIP. Similarly, George Moore, vice president of sales and marketing at StarVox Inc., believes that network administrators are "absolutely excited" about VoIP.
In contrast, Thomas Nolle, president of CIMI Corporation, a technology assessment firm, paints a picture of disinterest. To illustrate his point, Nolle describes an April 1998 conference during which he addressed a group of corporate information officers (CIOs) who represent 100 of New York's largest companies. Nolle asked these CIOs if they were carrying voice over their IP networks or if they would be in the near future. According to Nolle, not one of these CIOs answered in the affirmative. In fact, Nolle reports that one CIO said he felt no need to risk his business by subjecting it to the "unknown voice quality" associated with VoIP.
Nolle's informal survey supports Sara Hofstetter's hypothesis that "only two or three Fortune 1000 companies have implemented VoIP on a corporate level." Hofstetter is the director of public relations for Integrated Device Technology (IDT) Inc. Despite this hypothesis, however, Hofstetter believes that VoIP is worth getting excited about. As Hofstetter explains, VoIP "is a fantastic opportunity" for companies.
With inconsistent responses such as these, you may not know what to think about VoIP. Is VoIP a great opportunity or a great risk for your company? Because you don't know the answer to this question, you may be justifiably wary about VoIP--but you should still be interested in VoIP. Why? Because some, but not all, VoIP solutions are sound solutions for corporate networks--solutions that deliver good voice quality and promise significant cost savings. However, whether VoIP solutions are defensible options for corporate networks depends on the type of voice communications you're talking about and the type of network over which you plan to carry voice. What you need to know is which types of VoIP solutions you should check out--and which ones you should chuck out. (For a list of VoIP solution providers, see "VoIP Equipment Vendors and Service Providers.")
VOICE OVER THE INTERNET--THE SOUND AND THE WORRY
VoIP is industry jargon that refers to the ability to digitize and compress voice and then compile it into packets for transport over a public or private IP data network. Some people broaden the scope of VoIP by using it to include the transport of streaming audio files, which you can download from an intranet or the Internet. In contrast, other people (including a few members of the press, some VoIP service providers, and a handful of industry analysts) unintentionally restrict the scope of VoIP. These people use Internet telephony and VoIP interchangeably, implying that VoIP refers only to the transport of voice over the Internet. Generally, however, VoIP refers to the transport of real-time conversational voice (or fax) over a private intranet, a publicly managed IP network, or an unmanaged public IP network, namely, the Internet.
Most people agree that carrying voice over managed public or private IP data networks is a viable corporate alternative to maintaining separate voice and data networks. (A few people do argue rather loudly against this idea.) However, most people also agree that carrying voice over the Internet is not a viable corporate alternative.
With the exception of one or two useful voice-over-the-Internet applications, the Internet, most people agree, is not ready now, nor will it ever be ready, to carry voice traffic for companies. Companies demand toll-quality voice, that is, voice that sounds like voice over the Public Switched Telephone Network (PSTN). In other words, companies demand a quality of voice that the Internet simply cannot provide.
Even die-hard VoIP advocates such as Jeff Pulver, president of an Internet-based consulting firm called pulver.com, freely admit that the Internet will never make it as a corporate alternative to the PSTN: "Whoever said that [the Internet] had to?" asks Pulver in exasperation. Widely recognized as an expert in the VoIP industry, Pulver explains that companies would be remiss to bother with VoIP over the Internet. "Quality of voice over the Internet today may be the best you'll ever hear it," Pulver says, quickly adding that, at best, "that quality is predictably unpredictable." Pulver maintains that it "makes no sense" to put a company at risk by subjecting its voice traffic to "an environment where you can't control the quality." Pulver concludes, "Voice over the Internet is best to sold to people who don't care about quality." (For more information about pulver.com, Pulver's company, visit http://www.pulver.com.)
Is Predictably Unpredictable Quality Acceptable for Fax?
Despite its limitations, the Internet is not totally useless as a corporate telephony vehicle. For example, companies can plausibly use voice over the Internet to enhance their World-Wide Web sites. (For more information about enhancing your company's web site with VoIP applications, see "Give Your Company's Web Site a Voice.") In addition, companies can use VoIP applications, such as PSINet's InternetPaper, to route fax transmissions over the Internet.
However, whether or not the Internet provides a suitable environment for fax transmissions is open to dispute. Some VoIP equipment vendors and VoIP service providers, including Nortel Networks and Delta Three, believe that faxing over the Internet is a legitimate option for companies. However, other VoIP equipment vendors, most notably Lucent Technologies, clearly disagree.
In a white paper titled "The Market for Internet Telephony," Delta Three actually implies that faxing over the Internet is better than faxing over the PSTN. (You can download this white paper from http://www.deltathree.com/company/press/whitepaper.asp.) As Delta Three reports, "some 28% of outbound international faxes from the United States failed on the first transmission attempt." With Internet-based faxing, on the other hand, the VoIP gateway closest to the recipient can try the transmission repeatedly until that transmission succeeds, "meaning that the sender need only stand at the fax machine once." ("The Market for Internet Telephony," p. 2.)
In contrast, in a white paper titled "Impact and Performance of Lucent's Internet Telephony Server (ITS) Over IP Networks," Lucent Technologies implies that the Internet is worse for fax transmissions than it is for voice traffic. (You can download this white paper from http://www.lucent.com/enterprise/internet/its-e/documentation.html#white.) According to Lucent's white paper, "network performance requirements for fax over IP [are] more stringent than [the requirements for] voice over IP." (P. 7.) Whereas a one-way packet delay of up to 400 milliseconds is tolerable for voice traffic, a one-way packet delay exceeding as few as 300 milliseconds is "unacceptable" for a fax transmission. (Pp. 8-9.) (Lucent also notes that for both voice and fax, delays of fewer than 200 milliseconds are preferred.)
Despite the discrepancies in the claims of VoIP equipment vendors and service providers, applications for transmitting fax over the Internet might be worth looking into--particularly if a considerable portion of your company's phone bill is for international faxing charges. For example, if your company's phone bill rivals phone bills for Fortune 500 companies, you might want to look at Internet telephony alternatives to faxing over the PSTN. According to Pulver, 40 percent of the charges on phone bills for Fortune 500 companies are charges for international fax transmissions. Consider how much your company could save if every one of those international fax transmissions were free (after the initial cost of the Internet fax application, of course).
A Few Extra Digits to Save a Few Extra Pennies
Although the Internet is not ideally suited for carrying voice and fax traffic, most people in-the-know agree that public IP networks and private intranets are very well suited for carrying voice and fax. A wide range of VoIP solutions for public and private IP networks are available, but not all of these solutions are appropriate for your company.
For example, several next-generation telephone companies (telcos), including Delta Three, Qwest Communications, and IDT, are building their own IP networks over which customers throughout the world will eventually be able to exchange voice, data, and even video. These telcos currently offer VoIP phone-to-phone services, such as Qwest Communications' Q.talksm and IDT's Net2Phone Direct. These long-distance services route voice and fax traffic over a managed IP network, rather than the PSTN, and purportedly save money. To use these services, however, you must dial into the telco's IP network before dialing the number you are trying to reach.
Although the exact steps vary between services, placing a call using one of these services would involve the following basic steps:
You dial into the service provider's network using a local access number, a long-distance number, or a toll-free number.
You enter a personal identification number (PIN) or an account number.
You dial the number you are trying to reach (including the area code).
Depending on the size of your company and the price breaks your long-distance carrier offers, this type of service can save your company money. For example, such solutions might appeal to small companies because they typically don't get the same price breaks from long-distance carriers as larger companies get. Hence, the rates the next-generation telcos offer might be better than your company's current rates. IDT, which controls nearly 30 percent of the IP telephony services market, charges as few as 5 cents per minute for state-to-state calls that you place using its Net2Phone Direct services and as few as 10 cents per minute for international calls.
On the other hand, this type of service might be impractical for your company, regardless of its size. After all, your company can reap the savings these services offer only if employees use the service. So before researching your options, ask yourself this question: Could you really train employees to consistently dial all of those extra digits before dialing the number they are trying to reach?
GATEWAYS BREAK THE SOUND BARRIER
One of the most practical VoIP solutions for companies of any size is a gateway solution. VoIP gateway solutions enable you to place calls between branch offices for free or for exceptionally low rates--after the cost of the gateway solution itself, of course. If your company already has an intranet, you might opt to purchase a gateway solution from a VoIP equipment vendor, such as Nortel Networks or StarVox.
Nortel Networks' V/IP Phone/Fax IP Gateway offers small- to medium-sized companies a way to route voice and fax traffic between offices over their private intranets. You can install the V/IP gateway on a 486 or above server running NetWare 3.12 or NetWare 4.x, or you can install the V/IP gateway on another supported operating system, such as Windows 95.
The V/IP gateway is the only VoIP gateway available today that runs on NetWare servers. (However, Lucent Technologies plans to make its IP ExchangeComm System support NetWare 5 soon.) In addition to being connected to your company's intranet, the V/IP gateway connects to your Private Branch eXchange (PBX) or key telephone system via analog and digital voice interface cards (VICs).
StarVox's StarGate Server is best suited for companies with at least 10 offices, particularly if some of those offices are in other countries. Like the V/IP gateway, StarGate Server enables you to route telephone calls between sites over your company's intranet links.
StarGate Server runs on Windows NT 4.0 and, unlike the V/IP gateway, can extract data from Novell Directory Services (NDS), which Moore describes as "the best directory available today--bar none." StarGate Server is the only VoIP gateway available today that integrates with NDS. (However, Lucent Technologies plans to integrate its IP ExchangeComm System with NDS soon.) Like the V/IP gateway, StarGate Server connects to both your company's intranet and to the PBX or key telephone system.
If your company does not have an intranet, you might opt to purchase a gateway solution from a VoIP service provider, such as IDT. IDT offers companies a relatively new service called Project David, which enables you to route voice, data, and video traffic between branch offices over IDT's managed IP network. To subscribe to Project David, you must first purchase and install IDT's proprietary VoIP gateways, which include hardware and Unix-based software. IDT manages these gateways 24 hours a day, seven days a week. You must also lease lines to IDT's IP network from each of the branch offices where you install an IDT gateway.
Voices Carry--But How?
Gateway solutions vary widely in terms of how each is implemented, how each works, and what each offers. However, the following example gives you a general idea of how a VoIP gateway solution works:
Suppose that you implement a VoIP gateway solution on your company's intranet, which connects the corporate headquarters located in Iowa to branch offices located in several other states and countries, including Japan. (See Figure 1.) This solution enables phone-to-phone, PC-to-phone, and PC-to-PC VoIP communications between any user, anywhere on your intranet. You also install a VoIP gateway at each branch office and attach each of these VoIP gateways to both the intranet and the PBX or key telephone system in each office.
Figure 1: A basic VoIP gateway solution
Now suppose that Joe Schmoe in Iowa wants to call Lee Schmee in Japan. To do so, Joe simply picks up his phone and dials a number, such as a 6 or an 8, indicating that this call should be routed through the local VoIP gateway. (Dialing the extra digit shouldn't strike Joe--or you--as odd. You probably use or have used a PBX system that requires you to dial a 9 or some other digit before dialing an outside line.) Upon hearing a second dial tone, Joe then dials Lee's number.
The PBX in the Iowa office intercepts Joe's call and forwards this call to the VoIP gateway, which is also in the Iowa office. Next, the Iowa VoIP gateway finds the IP address of the remote VoIP gateway that is associated with Lee's telephone number. In this case, the remote VoIP gateway is the gateway at the branch office in Japan. The two gateways then establish a connection, Joe hears the line ringing, and (for the sake of this discussion) Lee answers. This telephone call will (or at least should) seem perfectly ordinary to both Joe and to Lee. From Joe's perspective, he dialed, the phone rang, and Lee answered. From Lee's perspective, the phone rang, she answered, and she and Joe started talking. Neither Joe nor Lee are aware--or even care--about the following facts of their phone call:
For each word Joe and Lee spoke, each of the VoIP gateways making this call possible used a codec (which stands for voice coder/decoder) to convert the analog signal their voices generated into digital signals.
The VoIP gateways also used codecs to compress the digital voice signals and break them into UDP/IP packets. (For information on why VoIP applications use UDP rather than TCP, see "UDP: The Choice for Voice.")
Joe's and Lee's voices were transmitted over their company's intranet, which Joe and Lee also used to view and discuss a shared document during their conversation.
Because the call was routed over the company's intranet, the call was essentially free. (For more information about the cost and potential savings of VoIP gateway solutions, see the "Sounds Like Money In The Bank" section.)
Upon receiving packetized voice signals, each VoIP gateway used a codec to reconvert these digital signals into analog form before sending them to the locally attached PBX.
During Joe's and Lee's conversation, the VoIP gateways sent between 20 and 40 voice packets per second. (Lucent Technologies estimates that during a typical phone conversation, a gateway sends 20 to 40 voice packets per second. See "Impact and Performance of Lucent's ITS over IP Networks," p. 2.)
Neither Joe nor Lee would know or care about any of these details because the call would sound just like a call routed over the PSTN. More specifically, any differences between the sound of the VoIP call and a traditional call would be imperceptible--at least, that's what VoIP gateway vendors and service providers claim.
Keep Your Ears Open
Of course, not everyone accepts these claims. For example, NetWare Connection asked Nolle what advice he would give network administrators who are considering the purchase of a VoIP gateway. In response, Nolle warned that network administrators better make sure that everyone in their company "is aware of the impact that [a VoIP gateway] will have on the quality of speech and on the performance of the data network." Nolle added that network administrators "better be sure that [their] savings will be accrued very quickly to pay off the cost" of the gateway.
Nolle's implication is clear and stands in stark contrast to claims of VoIP equipment vendors and service providers: VoIP gateways, Nolle believes, negatively impact voice quality, consume a lot of bandwidth, and cost a lot of money. Although Nolle's implication is arguable, his advice is not. Both VoIP equipment vendors and service providers would agree with Nolle: When you check out a VoIP solution, you should look for a solution that offers good voice quality, low-bandwidth consumption, and enticing cost savings.
In fact, VoIP equipment vendors and service providers believe that these evaluation criteria are essential to the success not only of particular VoIP solutions but also of the VoIP industry in general.
THAT VOICE (OVER IP) SHOULD HAVE A FAMILIAR QUALITY
Unlike residential users, who are willing to settle for poor voice quality if they can finagle a free long-distance phone call over the Internet, corporate users won't trade quality for cost savings. If you implement a VoIP solution to carry voice over your company's intranet and the voice quality isn't precisely what users are accustomed to hearing, you might be the only one using that nifty solution (you and a few corporate devotees in accounting). The bottom line is that corporate users demand good voice quality--that is, voice that sounds like the voice they're used to hearing over the PSTN.
Unfortunately, IP networks have a few quirks that are tolerable when transporting data but can become annoying or, worse, intolerable when transporting voice. For example, packet delay and jitter are common on IP networks.
Packet delay refers to the average length of time it takes a packet to traverse an IP network. Delays occur on IP networks for several reasons. For example, delays occur because there are limits to how fast signals can be propagated over physical links and because network devices, including codecs and routers, have to process these signals, and processing takes time.
Network congestion also contributes to packet delay: Signals queue up to await processing at the routers they encounter during their journey through the IP network. The more congested the network, the longer the queue.
Variable delays, such as those associated with queues and processing, cause packets to arrive at different times, an effect called jitter. Jitter can negatively impact voice quality. In fact, if voice packets that comprise a conversation reach a codec at dramatically different times and the codec forwards these packets to the listener in the order they arrive, jitter could turn a conversation into gibberish.
When a network transports data, packet delay and jitter are no big deal. But these seemingly harmless quirks can negatively affect the quality of voice on an IP network. In fact, many industry analysts, equipment vendors, and service providers suggest that the Internet's long delays and high jitter are to blame for its inability to satisfactorily carry corporate voice traffic.
In marketing materials, VoIP equipment vendors and service providers state or imply that they can compensate for effects such as jitter, claiming that they provide the voice quality companies demand. For example, IDT claims that its Project David services provide "toll-quality transmission." Likewise, StarVox reports that the voice quality StarGate Server delivers is "indistinguishable from that of conventional PSTN telephone calls." Similarly, Nortel Networks' Tyre says that delay and jitter "are easy problems to overcome." Tyre adds that Nortel Networks has "been offering what we consider business-quality voice for the last couple of years with our V/IP product."
These claims are justifiable. VoIP equipment vendors and service providers can counter the impact of jitter on voice by using a jitter buffer. A jitter buffer is part of a VoIP gateway's memory. Gateways hold voice packets in their jitter buffers for a specified length of time before playing these voice packets (that is, before forwarding them to the listener). Holding voice packets enables the gateway to begin playing these voice packets only when it can produce a steady stream of speech. In short, Tyre says, jitter buffers offer "plenty of opportunity to play voice back in a naturally smooth fashion."
That Codec Moment
Despite such optimistic claims, some people remain skeptical about the voice quality VoIP technologies can deliver. For example, to VoIP cynics like Nolle, the quality of voice carried over an IP network is just not the same as the voice you're accustomed to hearing with PSTN. "VoIP is immediately discernibly different than voice over the traditional PSTN," Nolle argues, boldly adding that "anybody who says otherwise is lying to you."
Nolle believes the cause of this ostensible degradation in voice quality, at least in part, is voice compression. Without voice compression, VoIP would be nearly impossible. After all, uncompressed voice consumes 64 kbit/s. To minimize the amount of bandwidth VoIP would otherwise consume, codecs compress voice. Unfortunately, compressing and decompressing voice require codec processing time. Processing time contributes to delays on the network, and delays can affect voice quality. Thus, ironically, the voice compression technology that makes VoIP possible also potentially degrades voice quality.
However, the codecs that compress voice are also designed to maintain an acceptable voice quality. In fact, standards for codecs include provisions for balancing voice compression and voice quality. During the past several years, the International Telecommunications Union (ITU) has developed seven standards for codecs: G.711, G.722, G.726, G.727, G.728, G.729, and G.723.1. (The ITU is a body within the United Nations Economic, Scientific, and Cultural Organization, more commonly known as UNESCO. You can find more information about ITU at http://www.itu.int.)
VoIP gateways commonly use the G.729 standard or the G.723.1 standard. For example, Nortel Networks' V/IP gateway uses a G.729 codec, and StarVox's StarGate Server uses a G.723.1 codec.
Although G.729 and G.723.1 offer different compression rates, they both purportedly deliver the same voice quality. More specifically, G.729 compresses voice to only 8 kbit/s. G.723.1 offers an even lower bit rate of between 5.3 and 6.3 kbit/s. Despite these low bit rates, Daniel Minoli, who has researched voice over data networks for more than 20 years, claims that both standards provide "good" voice quality. (See Daniel Minoli and Emma Minoli, Delivering Voice Over IP Networks, New York: John Wiley & Sons Inc., 1998, p. 155.)
THE SOUND OF SILENCE? CONSERVING BANDWIDTH
In addition to voice compression, the best bandwidth-reducing technology is silence compression (sometimes called silence suppression). Some codecs use silence compression algorithms to detect silences in voice conversations (or fax transmissions). These codecs conserve bandwidth by not sending packets at all during periods of silence. The codecs that IDT, Nortel Networks, and StarVox use in their VoIP gateways (Project David gateways, the V/IP gateway, and StarGate Server, respectively) all support silence compression.
As reported by Nortel Networks in a white paper titled "Voice/Fax Over IP," silence constitutes approximately 60 percent of a typical telephone conversation. (You can download this white paper in .pdf format from http://www.micom.com/WhitePapers/index.html.) Hence, silence compression can reduce voice packet bandwidth consumption by as much as 60 percent.
Nortel Networks estimates that because of the voice and silence compression technologies its G.729 codec uses, the V/IP gateway reduces the bandwidth voice packets consume to as few as 4 kbit/s. Nortel Networks adds that, over a 20- to 30-minute period, a remote-site VoIP gateway trunk is active less than 25 percent of the time. Thus, the net average WAN bandwidth consumption per VoIP gateway trunk over a 20- to 30-minute period is about .25 x 4 kbit/s, or 1 kbit/s. One kbit/s is only approximately 1.7 percent of a 56/64 kbit/s link. (See "Voice/Fax Over IP," p. 36.)
G.729 codec bandwidth 8 kbit/s IP router overhead 2 -7 kbit/s (2 kbit/s with router header compression; 7 kbit/s without router header compression) Total Bandwidth 10-15 kbit/s Less 60 percent silence 6-9 kbit/s Net codec bandwidth consumption 4-6 kbit/s averaged over a 20-30 second period
Using silence compression, a G.729 codec can cut its bandwidth consumption in half. (This information is taken from Nortel Networks' "Voice/Fax Over IP," p.36. You can download this white paper from http://www.micom.com/WhiterPapers/index.html.)
Despite saving a sizable chunk of bandwidth, silence compression does have a few potential drawbacks. For example, if the silence compression algorithm doesn't immediately recognize an increase in audio energy, one or more of the first words uttered after a silence might get clipped off. However, advanced silence compression algorithms don't have this problem.
Another potential problem is that when no packets are being sent, you hear an eerie silence--the type of silence that prompts you to ask "Are you there?" Sophisticated silence compression algorithms avoid this dead-line effect by inserting comfort noise, or white noise, that lets you know the other party didn't hang up on you. StarVox's StarGate Server inserts this type of comfort noise during periods of silence.
More sophisticated silence compression algorithms include the ability to sample background noise and regenerate that noise during periods of silence. Nortel Networks' V/IP gateway and the gateways IDT uses for its Project David services both use background noise regeneration algorithms.
SOUNDS LIKE MONEY IN THE BANK
Because of concerns about voice quality and concerns about bandwidth, you may wonder why a company would bother with VoIP. Ask a VoIP equipment vendor, service provider, or industry analyst what attracts companies to VoIP, and you'll get the same answer: "What's driving [companies] to [VoIP] now is the potential to save money," says StarVox's Moore. In his September 1, 1998 "Pulver Points on the Internet Telephony Industry," Pulver says that VoIP "is all about cheap minutes. This is not hype," Pulver says, "but fact." (You can view these and other editions of "Pulver Points" online at http://www.pulver.com/points/index.html.)
Discussing the cost and potential savings of any technology is difficult. However, a few examples might help you form a general idea about how cheap VoIP minutes really are. Suppose your company purchases a provider's VoIP services, such as IDT's Project David services. Your company's initial investment will include the cost of the gateways for each office and the cost of the connections from these offices to IDT's IP network. Hofstetter says that the cost of each gateway depends upon its size and complexity. For example, a one-port, simple gateway costs about U.S. $200, whereas a 32-port fully integrated, sophisticated gateway costs about U.S. $40,000.
The cost of each connection from your company's offices to IDT's network depends on the type of connection your company needs, from a 56 kbit/s line to a T1 connection. Thus, the cost of each connection could range anywhere from U.S. $500 to $1,000 per month.
Once your company's gateways and connections are in place, your company starts saving. Although IDT will charge your company for each call made over its IP network, these charges will probably be considerably less than your company is paying now. For example, for its Project David services, IDT charges only 3 cents per minute for domestic calls and as few as 9 cents per minute for international calls.
The initial cost of a gateway solution that you purchase from a VoIP equipment vendor is probably going to be more than the initial cost of a gateway solution from a VoIP service provider. However, after you purchase and install a gateway from an equipment vendor, intracompany phone calls are free. (Your company must still pay for calls to persons outside of its intranet. To see how you can save on these calls as well, see "Off The Record: Off-Net Calling.")
For example, Nortel Networks has created a sample cost-savings scenario based on a 35-site network with V/IP Phone/Fax IP Gateway servicing 24 T1 digital channels at headquarters and one V/IP Phone/Fax IP Gateway at each site servicing an analog dual-channel. The initial cost for this solution is U.S. $76,720. Nortel Networks estimates that this company would spend about U.S. $554,400 in voice and fax charges, based on 10-cents per minute charges over five years. Given this estimate and the immediate, hard-cost savings of toll-free, intracompany calls, the V/IP gateway could save this particular company U.S. $477,680 within five years. This company could expect to recover the cost of its initial investment in the V/IP gateway within 8 months. (To view the estimated cost savings per month, visit http://www.nwconnection.com.)
StarVox provides free, online estimates on the number of months it would take for you to recover the undisclosed costs of StarGate Server and the estimated dollar amount you would save yearly by using StarGate Server. To get an estimate, complete StarVox's cost-savings worksheet, which is posted on StarVox's web site at http://www.starvox.com. Click the Network Telephony Theater selection from the Menu, and choose Start Saving Now from the list that appears.
StarVox bases its cost-savings estimates on variables such as the number of corporate users, the number of connected sites, the percentage of users that are domestic, and the cost per minute your company pays now for long-distance domestic and international calls. For example, the default answers to the worksheet questions show that a company with 20 sites and 5,000 corporate users, 90 percent of whom are domestic, could expect to recover the cost of the initial investment in StarGate Server within 6.65 months. (This example assumes that the company was paying U.S. 8 cents per minute for domestic long-distance calls and U.S. 50 cents per minute for international long-distance calls before implementing StarGate Server.)
StarVox also suggests that the yearly savings for such a company would be U.S. $1,873,267.20. If that amount strikes you as inordinately high, it's because StarVox bases dollar savings not just on the hard-cost savings associated with bypassing tolls incurred on the PSTN but also on savings associated with "soft costs," such as gains in user productivity.
For example, StarVox estimates StarGate Server can save users who make about 40 calls per day an estimated .33 hours per day, which adds up to 66 hours saved per user, per year. If the average hourly salary of those users is U.S. $27 and there are 1,000 users, the savings per year equals approximately U.S. $1,777 per user, or $1,776,720 total in productivity gains for the year. (For more information about how StarVox estimates the cost savings associated with gains in user productivity, visit http://www.nwconnection.com.)
Are these estimates legitimate? After all, how can a VoIP gateway help users be more productive? In fact, most VoIP gateways don't. But StarGate Server offers more features than most VoIP gateways. According to StarVox, StarGate Server is not simply a VoIP gateway: StarGate Server is a network telephony server. With a network telephony server, Moore explains, you get more than just free phone calls.
Moore describes other VoIP gateways as "dumb pipes" because they leave behind many features corporate users expect. "With conventional telephony," Moore points out, users could "transfer a call, conference a call, or do call forwarding." Yet most VoIP gateways don't offer such conveniences. StarGate Server provides these PBX features. (For more information about StarGate Server, see "NDS-Enabled Applications: What Do They Have That Other Applications Don't," NetWare Connection, Aug. 1998, pp. 12-14. You can download this article from http://www.nwconnection.com/aug.98/nds.)
IN THE MARKET FOR A NEW SOUND SYSTEM?
Are you interested in VoIP yet? Predictions regarding the increase in VoIP-related revenue suggest that even if you aren't interested now, you will be interested within the next two to five years. For example, according to Frost ' Sullivan, as reported by Delta Three, revenues for the VoIP gateway market are expected to increase at a compound annual growth rate of 229 percent, reaching U.S. $1.81 billion by the end of 2001. (See "The Market for Internet Telephony.")
Pulver estimates that VoIP equipment sales alone will exceed two billion dollars by 2001. By 2003, Pulver predicts, "20% of all voice traffic around the world will be IP-based."
IDT's predictions are more optimistic than Pulver's. According to Hofstetter, IDT believes that within the next two to three years, IP networks will carry not only voice but also data, cable, and video traffic.
Even cynics like Nolle admit that "five years from now, there will probably be a reasonable market for VoIP services." Nolle, like many others (including Pulver), believes the VoIP market will develop as an adjunct to the Virtual Private Network (VPN) market.
Despite these predictions, any rumors you might have heard about the PBX being dead or about IP-based networks killing the PSTN have been greatly exaggerated. The PBX and PSTN are not dead now nor will they ever be. In fact, in an August 4, 1998 press release, Forrester Research Inc. concludes "that the dream of a single network infrastructure for corporate voice and data will never be fully realized." (You can download this press release from http://www.forrester.com/Press/Releases/Standard/0,1184,29,00.html.)
Although it seems probable that there will always be separate voice and data networks, it seems equally probable that placing voice and data traffic onto one IP-based network will become increasingly common. The August 1998 press release from Forrester Research reports that of the Fortune 1000 corporations Forrester interviewed, "more than 50% expect to migrate some voice traffic onto their data network within two years."
Which raises a new question: When will you migrate some voice traffic onto your company's IP network?
Linda Boyer Kennard works for Niche Associates, which is located in Sandy, UT.
VoIP Equipment Vendors and Service Providers
Equipment Vendor or Services Provider
|
Telephone Number
|
Delta Three http://www.deltathree.com |
+44-1483-457609 Europe +571-312-1846 Latin America +61-2-95511700 Asia Pacific +972-2-649-1222 Israel 1-888-DELTA-30 U.S. and Canada |
IDT's Click2Talk Division http://www.click2talk.com |
1-201-907-53697 1-888-872-1230 U.S. and Canada |
1-201-928-2990 1-800-345-7015 U.S. and Canada |
|
Lucent Technologies http://www.lucent.com |
1-908-582-3000 |
Nortel Networks http://www.micom.com |
1-805-583-8600 1-800-642-6687 U.S. and Canada |
PSINet http://www.psi.ca |
1-416-228-3400 |
Qwest Communications International Inc. http://www.qwest.net |
1-800-860-2255 U.S. and Canada |
StarVox Inc. http://www.starvox.com |
1-408-383-9900 |
Give Your Company's Web Site a Voice
"Regardless of how sophisticated your web site may be," Integrated Device Technology's (IDT's) Click2Talk web site reads, "razzle-dazzle graphics are no substitute for human contact when it comes to closing a sale." (You can access IDT's Click2Talk World-Wide Web site at http://www.click2talk.com.) Human contact between customers and your company's sales representatives is precisely what IDT enables with Click2Talk.
Click2Talk is one of a new breed of VoIP applications that give your company's web site a voice. If you add the Click2Talk icon to the web site, visitors can click that icon to speak--over the Internet--with one of your company's sales representatives. A visitor needs only a multimedia computer and IDT's Net2Phone software, a PC-to-phone application. (You can download free Net2Phone software from http://www.net2phone.com.)
Your company's sales representatives need only telephones. They don't need to be online. In fact, they don't even need a computer. Yonah Lloyd, director of IDT's Click2Talk division, explains that Click2Talk provides "real-time voice communication that originates online from any PC anywhere in the world but terminates on a standard telephone."
The value of Click2Talk lies in its potential to secure sales by enticing two types of customers your company might otherwise lose:
Customers who are reluctant to enter personal, financial information on the Internet
Customers who are reluctant to pay for international calls to purchase something
According to Lloyd, a company might be losing sales as a result of people's fear of the Internet. "Most people don't want to throw around personal information, especially credit card information, on the public Internet," Lloyd suggests. Lloyd's implication is ironic but true: Most people feel more comfortable giving their credit card information verbally than they feel giving that information in writing--even though that information, whether verbal or written, is carried over the same lines, in this case, Internet lines. Thus, when faced with entering credit card information to complete a purchase online, many potential customers leave the web site and your company loses the sale. But if these would-be customers could click an icon to speak with an actual person, they might just stick around long enough to complete the purchase.
Expensive international calls might also create barriers between your company and its potential customers. Using Click2Talk, overseas customers can place a free phone call to purchase something they see on your company's web site.
Adding the Click2Talk icon to your company's web site, according to Lloyd is "a no-brainer. . . . It takes about one minute." After you add the icon, the rest is up to your customers. What happens after a potential customer clicks your Click2Talk icon depends on whether or not that customer already has IDT's Net2Phone.
If a customer who already has Net2Phone clicks the Click2Talk icon, "the Net2Phone software automatically dials the preprogrammed phone number," Lloyd explains. Within seconds, that customer is talking to one of your company's sales representatives. However, most Internet users do not have Net2Phone--not yet anyway. IDT is working to ensure that more people get Net2Phone. In fact, "within the next six to 12 months," Lloyd projects, "pretty much anyone buying a computer from most of the major manufacturers [in the U.S.] and around the world will have Net2Phone."
If a customer does not already have Net2Phone, he or she will have to download Net2Phone before using Click2Talk. Companies that add the Click2Talk icon to their web sites include a line of text that says something to this effect: "In order to use Click2Talk, you must have Net2Phone. If you do not have Net2Phone, click here."
When potential customers click on the hypertext link, Net2Phone is automatically downloaded, free of charge and pretty quickly. The Net2Phone package is approximately 1.3 MB, which takes about five to six minutes to download for users with only 28.8 connections. Once the customer has downloaded Net2Phone, he or she can click the Click2Talk icon to speak with one of your company's representatives.
As the main article points out, most people agree that the Internet is not up to corporate standards in terms of voice quality. Not surprisingly, the quality of Click2Talk voice traffic doesn't exactly rival the quality of calls over the Public Switched Telephone Network (PSTN). In fact, the quality of Click2Talk voice is "probably similar to cell phone quality," admits Lloyd without apology. When you use Click2Talk, Lloyd continues, "You know for sure that you're not talking to somebody on a full standard phone, but you can still have a conversation." However, Click2Talk users are home users, who are far less concerned about voice quality than corporate users.
According to Lloyd, more than 100 companies are already using Click2Talk, including Andrea Electronics (http://www.andreaelectronics.com/product.htm), 1-800-FLOWERS (http://www.1800flowers.com/flowers/help/clicktotalk.asp) and Sheraton Boca Raton Hotel (http://www.sheratonboca.com). You may want to visit one of these web sites and try out Click2Talk yourself.
UDP: The Choice for Voice
Like all packets on an IP network, voice over IP (VoIP) packets have either TCP/IP or UDP/IP headers, depending on which protocol is being used to transport the packets over the IP network. Whether VoIP packets use UDP or TCP as the transport protocol depends in part on whether the signals these packets are carrying are voice or fax signals.
For fax signals, VoIP gateways append either TCP/IP or UDP/IP headers to the packets that carry those signals.
For voice signals, VoIP gateways append only UDP/IP headers to the packets that carry those signals.
According to a Nortel Networks white paper titled "Voice/Fax Over IP," UDP is the protocol of choice for transporting voice over IP for several reasons, including the following. (You can download this paper from http://www.micom.com/WhitePapers/index.html.)
Unlike TCP, UDP doesn't require that the sending and receiving devices do any handshaking. The handshaking that TCP requires causes unnecessary delay, which can affect voice quality. UDP requires no such handshaking and, therefore, causes less delay. (See "Voice/Fax Over IP," p. 16.)
Unlike TCP, UDP doesn't retransmit data when it detects an error. When TCP detects corrupted packets, it retransmits those packets. Retransmitting voice packets delays the delivery of those packets and can negatively affect voice quality.
With UDP, a corrupted packet is usually thrown out, and the last good packet simply replayed. Listeners seldom notice packet replay if replayed packets comprise less than 5 percent of the packets transmitted during a given session. (See "Voice/Fax Over IP," p. 16.) On private networks, packets are seldom corrupted and, therefore, replaying packets is seldom necessary.
Off The Record: Off-Net Calling
If you purchase and install a voice over IP (VoIP) gateway solution from an equipment vendor, calls between company offices will be free. However, calls to your partners and customers, who are not on your company's intranet, will not be free. These calls will go over the Public Switched Telephone Network (PSTN) and, consequently, accrue whatever costs your company's long-distance carrier is charging--unless you engage in a little off-net calling.
Off-net calling enables you to use your gateways to avoid some of the charges you would otherwise have to pay a long-distance carrier for calls outside your company's intranet. Off-net calling works like this: Suppose you have a VoIP gateway at corporate headquarters in Iowa and a gateway and PBX at a remote site in Japan. Now suppose you want to call a partner, who is is located in Japan and is not on your company's intranet. To avoid some of the costs you would otherwise incur by calling this partner directly, you could dial 6 for the local gateway, then dial the number for the remote gateway, then dial 9 for the PBX in Japan.
The next dial tone you heard in Iowa would be the dial tone of the PSTN in Japan. You could then dial the number of your partner in Japan, and the call would be a local call (or, at least, not an international call). A PBX that you use to sneak around charges for international, long-distance calls is called a "leaky PBX." A leaky PBX, according to Nortel Networks, is tolerated by phone companies and legal "in most countries." ("Voice/Fax Over IP," p. 26-27. You can download Nortel Networks' white paper in .pdf format from http://www.micom.com/WhitePapers/index.html.)
* Originally published in Novell Connection Magazine
Disclaimer
The origin of this information may be internal or external to Novell. While Novell makes all reasonable efforts to verify this information, Novell does not make explicit or implied claims to its validity.