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Voice-over-IP Across the Enterprise Network
Voice-over-IP technology first created a buzz with the arrival of Internet telephony. Consumers got excited by the prospect of using a PC and an Internet connection to dial up friends anywhere in the world and talk for hours without ringing up long distance charges. Never mind that the products were proprietary or that the quality had more in common with tin cans and string than a digital dialogue--the possibility of long-distance calls at local rates was enough to heat up the market. Companies of all sizes have since unleashed a flood of products, from PC software for end users to VoIP-PSTN gateways for carriers.
This sudden expansion of the market has resulted in substantially improved quality, raised the level of audio fidelity and strengthened support for industry-standard protocols, such as the ITU-T's Recommendation H.323. Thus fortified, VoIP technology is beginning to carve a niche in corporate networks. The question is, is it really ready to make this leap?
After giving VoIP technology a tryout across Network Computing's own distributed network, we're convinced that it's a bit premature to roll it out across an entire corporatewide enterprise network. Concerns about interoperability, security and bandwidth management are creating static on the line between VoIP and widescale deployment.
For example, while we managed to coax equipment from several vendors to interoperate at a very basic level, we could do so only by using the G.711 codec. But this generated tremendous utilization across our frame relay and ISDN networks, resulting in periodic signal loss, particularly when other traffic was introduced to the network. On top of that, our attempts to use features such as "hold" or "transfer" across vendors' product lines forced calls to drop. Although H.323 specifies that these features should be implemented, vendors are not yet doing so consistently.
There's good reason to believe these hang-ups will disappear over the next year or so. Vendors in this area will incorporate support for additional low-bandwidth codecs, and feature-implementation issues also are expected to be resolved.
But that doesn't necessarily mean you should wait until next year to dip your toes in the VoIP waters. While the technology clearly is not in shape for enterprisewide deployment today, it is eminently suitable for interoffice, long-distance, toll-bypass service, and even for isolated LANs that have the right infrastructure.
Segmenting the Technology
Every enterprisewide corporate telephone network has the same basic components, including end-user equipment (telephones, premises wiring) and back-end gear (PBXs, trunk lines). VoIP devices generally fall into these same two camps, with IP-centric equipment replacing analog handsets and wiring, and IP-based equivalents filling in for PBX and/or interconnect wiring.
Although most VoIP equipment today employs proprietary protocols, many vendors are beginning to support the ITU-T's Recommendation H.323 standard. This highly modular version of the H.320 multimedia-over-ISDN specification is tailor-made for packet-based networks. H.323 defines a variety of node types, the most common of which are identical to those in today's typical voice networks: terminals for the desktop, gateways for bridging the packet network to a standard telephone network, and gatekeepers that set up calls and provide other administrative services to the various devices.
H.323's modularity makes it extremely flexible, particularly for joining an existing voice network to VoIP equipment. This concept is illustrated in diagrams throughout this article. "Existing Voice Network" (top left) depicts a typical corporate telephone network composed of traditional analog technology; "Mixed Voice and Data Network" (bottom left) shows how you might replace some components of this network with H.323 components, while preserving other portions of your analog network. Finally, "Total VoIP Network" (below) illustrates the same network as it would appear with VoIP technology installed from end to end. Although products already are available that can bring this end-to-end VoIP network to life, they're not quite up to snuff. In fact, we strongly caution against trying to deploy VoIP end-to-end across your enterprise at this time.
Instead, we recommend limiting your VoIP implementations to a few key areas. Thanks to H.323's modularity, you can replace only select components on your network. For example, you might provide users in a new facility with VoIP equipment at the desktop, yet retain your existing PBX network at your corporate headquarters. Conversely, you could replace an outdated PBX cluster with IP-centric systems, while maintaining existing user-side equipment at the desktop.
Don't be hasty in your decision about where to implement VoIP, however. Every area of your network will be governed by individual factors that motivate (or discourage) the adoption of VoIP technologies. Each portion of your enterprise network has its own considerations and you have to treat each piece differently when planning your implementation. For instance, the opportunities to cut costs in remote offices are not the same as they are for local users. Similarly, bandwidth and infrastructure requirements for a large office or campus differ radically from those for a small office or a telecommuter.
To provide voice services over a digital network, you need to convert analog waveforms into packets of digital signals that can traverse the network. That's a job for codecs (coder/decoders) residing within all VoIP nodes on the network, including every end-user device and any gateways you might use. Unfortunately, because vendors have not yet implemented a common set of codecs, you will face interoperability problems with large-scale deployments.
H.323 specifies mandatory support for the G.711 codec--also known as Pulse Code Modulation (PCM)--a widely available codec used in many forms of digital telephony. But G.711 requires 64 Kbps of continuous bandwidth for every network end point. On a full-duplex voice circuit, a single 64-Kbps feed suffices, but on a packet-switched network such as IP, 128 Kbps of cumulative data is required if two users are speaking simultaneously.
The H.323 standard also specifies a laundry list of more efficient codecs that may be used. The two most popular implementations are G.723.1, which can use 5.3 Kbps or 6.3 Kbps for each end of the connection, and G.729, which uses 8 Kbps at each end. To complicate matters, some first-generation products support G.723.1 while others support G.729. So, to guarantee interoperability among different vendors' products you must use G.711 everywhere--and this means you must expect every call to consume 128 Kbps of continuous network bandwidth, or else you have to implement products from only one vendor.
Security is another major consideration. In version 2 of H.323, encryption and authentication are optional, though most implementations include no security protections at all. As a result, an H.323-aware network analyzer becomes an effortless wiretap. If you're on a shared-media network, anyone can monitor any conversation without ever leaving his or her desk.
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