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Data traffic is growing by leaps and bounds, but new opportunities in voice - yup, voice - networking are shaking up the market.
By David Rohde
Network World, 9/28/98
Ron Reaman knows something about compression. As manager of network communications at Copeland Corp. in Sidney, Ohio, he runs wide-area voice and data networks for the world's leading maker of compressor coils that go into refrigerators and air conditioners.
So when Reaman tells how he ducks monstrous international carrier tolls by compressing voice to 8K bit/sec for transmission to Thailand, you might want to listen up. Especially when it turns out he's using something you may already have in your data backbone: a 128K bit/sec AT&T frame relay permanent virtual circuit (PVC).
"The quality is very good," Reaman says. "And now we're looking at voice over frame relay to Hong Kong later this year." And on the drawing board for 1999: voice over frame relay to other parts of China, provided AT&T's international arm can get more frame switches into the country.
Couldn't he use Internet telephony instead of waiting for international frame relay capacity?
Think about calling into a noisy factory-floor environment over a jittery Web link. "To me, the quality isn't where I'd like it to be for a manufacturing environment," Reaman says.
Welcome to the world of voice networking 1998-style, where instead of wondering if you should integrate voice into the data network, it is simply a matter of figuring out which technology to use.
Seeking a break overseas
It's fair to say that voice over frame relay and voice over ATM are not exactly stirring up much enthusiasm in the canyons of Wall Street or the venture-capital corridors of Silicon Valley. There the buzzword is IP telephony, with much of the attention going to newly emerging carriers that are proposing to scrap circuit switching in favor of IP multimedia end to end.
And make no mistake: IP telephony is not a theory - it's actually being used. The question is, By whom and what for? The answer: mostly consumer residential carriers that are especially targeting members of ethnic groups in the U.S. who call family and friends in their native countries.
Some of these carriers are building global shared IP virtual private networks (VPN) to carry voice traffic without nailing up circuits. Others are doing what for many business users is unthinkable - just shipping the traffic out across the Internet, says Paul Wallner, president of Hypercom Network Systems, a supplier of multimedia network equipment.
Among these carriers are relatively unfamiliar names such as IDT, ITXC, RSL Communications and USA Global Link. Go to them and you might have a hard, if not impossible, time putting together a robust package of corporate voice and data services for a long-term contract, such as you can get from AT&T, MCI WorldCom or Sprint. What you will get in a heartbeat is a prepaid calling card of the type increasingly popular among consumers.
That's not to say some enterprises haven't found a place for international Internet telephony. Last year, Kanematsu USA, the U.S. subsidiary of a Japanese-based commodities trading company, installed an Internet Telephony Server-Enterprise (ITS-E) from Lucent Technologies to route voice calls to two ITS-E units in Tokyo and Osaka.
"At the beginning, the quality wasn't that great, although it was working," says George Emmett, assistant telecommunications manager at Kanematsu USA. "It sounded like it was bubbly because we lost a lot of packets. One day, one of the managers saw me pick up the phone and said, 'Are you on the Mickey Mouse phone again?' "
Nevertheless, with software upgrades, the ITS-E has been a real money-saver for Kanematsu. At the time it was installed, Kanematsu had a 64K bit/sec leased line from New York to Tokyo that could only support four voice-grade circuits.
Now a 24-port ITS-E is used to handle calls not only from New York to Japan, but also from Kanematsu's other U.S. offices to Japan. Callers in places such as Chicago and Los Angeles call the server in New York, which redirects their calls over the Internet, back across the continent and out across the Pacific to Japan. That way, Emmett explains, Kanematsu gets a trans-Pacific voice link for the price of a U.S. domestic long-distance phone call.
Domestic savings?
When it comes to domestic business telephony, users' widespread complacency about cheap rates is making it difficult for IP telephony carriers to build a business case. Even residential users now commonly get a rate of 10 cents per minute on nationally advertised circuit-switched calls from major carriers - 9 cents for users who agree to credit card billing.
IP telephony rates - which all seem to track what Qwest Communications charges for its Q.talk service, about 7.5 cents today - knock a little off those rates but are relatively meaningless to corporate users who routinely get 7 cents or less in negotiated contracts.
And the fast pace at which domestic IP telephony carriers are chewing up capital as they build out their networks means that their promises of lower rates achieved by low infrastructure costs may not materialize for some time.
You can't build a sustainable business case around the idea of offering cheap voice-over-IP telephone calls, says Karyn Mashima, vice president of data networking systems at Lucent. "In fact, right now, it costs the carriers more to provide voice over IP than traditional calls," she says.
Real savings
The real savings appear to be found not in switching from one carrier to another, but in getting rid of the separate voice carrier altogether. For users who have already moved to ATM backbones in the wide area, that's almost a no-brainer.
That's because key to ATM's ability to carry voice as well as data is its well-defined classes of service, such as constant bit rate or, more frequently, variable bit rate real-time circuits, which save even more money by directly interleaving voice cells with data cells.
Frame relay carriers have begun to emulate this setup by establishing classes of services or "priority PVCs" on their networks, with additional charges for PVCs guaranteed to have lower packet loss and delay characteristics. But it's a myth that voice over frame relay requires a special class of service from the carrier, according to some experts.
"It's our experience that a good public frame relay service can support voice, SNA and LAN traffic without separate classes of service or even separate PVCs," says Paul Wallner, president of Hypercom Network Solutions in Phoenix.
Prioritization can be achieved in the customer premises equipment (CPE) without ponying up extra dollars every month for souped-up carrier PVCs, Wallner says. In his view, most frame relay links tend to be underutilized most of the time, so it's pointless to add additional application-specific circuits when frame relay is supposed to save you money.
Still, loading voice onto the network presents a risk at those times when data traffic happens to spike. So you need to take certain precautions:
First, the broadly accepted G.729 standard for voice compression now calls for 8K bit/sec for good-quality voice. So your router or frame relay access device (FRAD) should be able to reserve 8K of bandwidth on the frame relay net for every simultaneous phone call from a particular location. "If you don't do this, a large file transfer can wipe out the voice traffic," Wallner says. If you anticipate more than four simultaneous calls, or 32K of reserved bandwidth, from a given branch, a 56K frame relay circuit may not be enough.
Any large data packet can wreak havoc on voice traffic, even if it's not within a burst. As a result, most frame relay CPE optimized for voice performs segmentation or fragmentation. For example, Hypercom's voice-enabled FRADs (V-FRAD) make sure that data traffic is not sent in packets larger than 256 bytes.
"Yes, you can put simultaneous voice and data on the same PVC using a V-FRAD, and that would be transparent to our network," confirms Melanie Hanssen, senior manager of data services marketing at MCI WorldCom in Richardson, Texas. But MCI WorldCom recommends that users acquire a separate PVC for voice traffic as an extra layer of protection, Hanssen says. MCI WorldCom is in the process of developing a managed V-FRAD service.
IP invades the PBXs
Another new option in voice networking is coming from PBX vendors and is best for users who are considering IP-based VPNs as an alternative to voice over frame relay. This option, called PBX IP trunking, takes advantage of the special call-routing features of a PBX system, such as hold, conference and transfer, while saving money by chopping out carrier tolls.
The most complete set of options developed so far is from Lucent, which by year-end will have a triple-threat PBX in place - one that gives network administrators a choice of ordinary time-division multiplexing (TDM), ATM switching and IP telephony all in one box.
Here's how it works: Lucent's flagship Definity Enterprise Communications System (ECS) originally was set up to deliver calls through TDM, generally via a single 64K bit/sec trunk or a 64K channel of a 1.5M bit/sec T-1 trunk.
Then, last year, Lucent added an optional ATM circuit pack for Definity's Center Stage, essentially the switching fabric of the PBX. That was followed this year by optional IP trunking, a 24-port card for the Universal Port Carrier shelf. Now users can have TDM, ATM and IP all in the same voice box.
Why use all three options? The answers center around scalability and feature transparency. The call-routing software in Definity carries two algorithms known as Automatic Route Selection (ARS) and Automatic Alternate Route (AAR).
The point of ARS and AAR is to let network administrators assign privileges and preferences to individual employees and workstations. Several years ago, that commonly meant that the CEO would be authorized to call around the world and get access to a trunk no matter what, while the phone on the factory floor would be limited to local calls. Today, that means network managers can theoretically decide which calls to route over a high-speed data WAN via ATM, which to offer up via IP and which to send over the public switched telephone network. If IP trunking is one of the three options utilizing ARS and AAR, the PBX software takes an additional step. It pings to determine the congestion on the IP network for each call, according to Marissa Russotto, Lucent's offer manager for Definity ECS. The purpose of the ping is to test for delay, jitter and packet loss. Users can set their own delay thresholds, but they can tolerate greater delay for fax than for voice, she says.
Don't forget voice mail
Many users who have no interest in IP telephony may unwittingly find that they get involved anyway through voice mail.
The Electronic Messaging Association (EMA) has approved a standard, called Voice Processing for Internet Mail (VPIM), that solves a major problem in the voice world: how to send messages over dissimilar systems. There's some urgency behind VPIM because voice has fallen behind data in the arena of interoperability. Almost anyone can send anyone else e-mail today over the Internet, but voice mail continues to be a welter of proprietary systems with almost no thought given to routing messages beyond one building or campus. VPIM uses the IP stack's Simple Mail Transfer Protocol and Multi-purpose Internet Mail Extensions protocols to encode voice messages exactly as if they were e-mail messages.
Here, Nortel and independent voice messaging vendor Applied Voice Technology have taken the lead over Lucent. Nortel introduced its Meridian Net Gateway earlier this year. Lucent is due to support VPIM on its Intuity Interchange platform only in mid-1999.
"VPIM is an IP telephony store-and-forward technology," says Bern Elliot, a Philadelphia-based independent consultant who is chairman of the EMA's Voice Messaging Committee. The wonderful thing about store-and-forward is that suddenly all the issues of latency and jitter go away, Elliot points out. That's because unless there's packet loss, the message will get delivered anyway, he says.
But there are some pitfalls. Voice messaging vendors need to avoid "clumsy user interfaces" in their VPIM implementations, Elliot says. A classic example is the Spoken Name feature. That's the part of most proprietary voice mail systems in which users, in set-ting up their mailboxes, speak their names in a separate input from the greeting so that they can be played back to callers who punch in their extensions.
The purpose of Spoken Name is simply to verify to the callers that they've selected the right mailbox. "In e-mail, it's pretty obvious who you're trying to reach because you have a keyboard," Elliot says. "In the voice world, it's not as obvious because you have a telephone keypad for your inputs."
So the next time someone touts voice over IP and you get the jitters about it, stop and think about what you already have in your corporate network. Between that frame relay carrier contract that's due to run for a while and the PBX and voice messaging systems you paid a pile of money for, you'll probably find safe, new packetized voice options long before Wall Street discovers you're never going to send your CEO's phone calls over the Internet.
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