Voice over IP (VoIP) is high on the agenda for businesses of all sizes for two reasons: to gain access to the high-profile, converged features that are difficult or impossible to achieve with traditional infrastructure, and to reduce call charges. Although there are other benefits to implementing some form of VoIP (such as resilient IP routing and flexible extension management), functionality and cost are clearly the two primary drivers.
Once the business decision has been made to implement VoIP, the next challenge is to work out how best to implement it. This depends on the business driver. If the main motivation is to gain features, then rolling out a completely new system to the desktop is the easiest way to get those benefits to the users, with an IP PBX, IP phones, and an upgraded LAN to carry the traffic.
The installation is relatively easy; there's no fiddling with proprietary PBX features or communication protocols to worry about and, because it's a completely new system, the IP system's functionality can be completely tailored to the business, including its office structure, applications, and call patterns. This means you needn't worry about matching up to users' existing expectations regarding call-handling features or voice performance.
However, even those businesses investigating VoIP for its features rather than its cost savings may be put off by the idea of a "rip and replace" implementation. Nobody can deny that businesses have made significant investments in their existing infrastructure, and it's unlikely that they'll be willing to write off these investments in the short term.
A common barrier to VoIP is the long-term contract that's tied to the PBX. The on-going maintenance is often agreed for up to a ten-year period, meaning that even if the PBX is removed, the business won't be saving on maintenance. Many businesses conclude therefore that they may as well continue to use it.
When implementing a complete VoIP solution, expenditure may actually rise. On-site managers, technicians and, significantly, the users themselves will need to be retrained to use their shiny new IP phones, IP PBX, and applications, rather than their familiar PBX functions. In addition to the quantifiable training costs, one must consider the lost productivity that results from such a significant change.
Over their evolution, PBXs have developed feature-sets that can only be described as comprehensive. Although IP-PBXs offer some clever capabilities and continue to evolve, they still lag in terms of sheer number of features. The missing features might be considered minor (advice of charge, for example), all it takes is for a user to notice that their oft-used feature is missing for the implementation to be tarnished.
Of course, there's the apprehension that every company feels about trusting their vital voice communication to a nascent technology. Will voice quality suffer? Will I get a dial tone? What if there's a power failure? It may not be enough to reassure customers that these issues can be avoided with a thorough design and careful LAN evaluation—it takes time for a new technology to become established and for its performance to stabilize.
This is doubly true in the case of voice infrastructure, where TDM PBXs are accurately described as carrier-grade. Businesses are unlikely to migrate their strategic and mission-critical call-center applications and call-routing to a technology they perceive to be buggy and unproven.
For many companies, the lure of VoIP's bells and whistles alone isn't enough to overcome the costs, risks, and sacrifices that a full VoIP implementation demands. However, there's another way for VoIP's flexibility, manageability, and cost savings to be felt without getting rid of the PBX or phones, and therefore without sacrificing features or familiarity.
One step at a time
The solution is to hide VoIP behind the customer's existing PBX, using a VoIP gateway to convert the TDM signals into IP signaling, and vice versa, as calls are routed in and out of the network (Fig. 1). As well as coding and decoding of IP signals into TDM voice to help the PSTN, PBX, and IP networks communicate, gateways provide call control and routing as well as sophisticated signaling, like audio-compression, echo-cancellation, silence suppression, and reporting.
1. Although the specific implementation varies, a typical configuration would look like the one shown, one with two PBXs and associated VoIP gateways.
On the telephony side, the gateway can offer ports compatible with European E1, domestic T1, and PRI-T1 connections. A host of signaling standards, including ISDN and Channel Associated Signaling (CAS), can also be supported, as well as QSIG and QSIG tunneling. For smaller sites and even homes, different capacity gateways are available equipped with digital or analog connectivity. Businesses don't just have investments in PBXs—the use of a gateway with a home-office worker's DSL line allows their existing office phone to be used as before, except the signal is transmitted over IP. The benefit of extending VoIP to home workers is that the phone can be treated as just another extension to the PBX.
Planning for successful implementation
When selecting a gateway, it's essential to not only plan for the correct number of voice channels required by users, but also to select a gateway that can scale easily to support future demands, and that can support multiple VoIP protocols and open industry standards. This scenario retains the old PBX and internal voice network, so you needn't worry about ensuring that an internal IP network is able to carry VoIP traffic. However, any IP link that carries time-sensitive traffic, such as voice, must offer sufficient performance and capacity to meet quality-of-service demands.
When implementing a gateway to interface between an internal PBX and an external IP network, you may not have control of the IP link—it may be owned by a service provider. But equally it may be an IP pipe between two business sites, say to deliver free inter-site calls. In this case, you must plan sufficient capacity based on the number of existing lines, users, and phones, and ensure that network elements are aware of different Classes of Service (CoS) to ensure Quality of Service (QoS) by minimizing jitter and latency. In either scenario, your gateway should keep performance levels up by quickly performing packetization, no matter what codec or how many lines are in use. If you can't ensure sufficient capacity, consider a more frugal codec, such as G.723.1, to reduce the amount of bandwidth each call requires (see the table).
Although the internal voice network, and indeed its inbound PSTN presence, can remain untouched, introducing VoIP to a site does present some security implications. For example, IP hardware can be affected by Denial of Service attacks. Minimize these by ensuring that the VoIP gateway and any other infrastructure elements remain behind a VoIP-aware corporate firewall for protection (Fig. 2). Secure gateways support NAT traversal to hop routing information over the firewall's IP range-change without disruption.
2. Minimize security breaches by ensuring that the VoIP gateway and any other infrastructure elements remain behind a VoIP-aware corporate firewall.
How it works
To connect multiple distributed PBXs as one unit over IP rather than costly leased lines, some gateways support the QSIG protocol. QSIG also supports advanced features not supported by VoIP, which lets the gateway make ancillary TDM infrastructure IP-ready without compromising functionality. Engineers should be aware that some PBXs don't play ball and use proprietary signaling, so thorough research and testing is essential before setting expectations for functionality and connectivity.
Typically, configuring the gateway to correctly route different types of calls to the PSTN or IP end-point, or to the right extension on the PBX, requires laborious construction of a routing table. The VegaStream Dial Planner operates dynamically, according to rules rather than a table, to ensure intelligent routing by number dialed, type of call, origination point, short code, prefix and so on. This maximizes flexibility and minimizes setup costs. Supporting DDI, the Dial Planner works standalone or can integrate with a separate gatekeeper or proxy server. Either way, there's no need to recode the PBX, and the Dial Planner accommodates multiple PBXs, for example, at multiple sites, because it can manipulate numbers as it routes them onwards. Configuring the system at the time of installation is easy because of this rules-based design, and because sophisticated reporting tools make testing fast and straightforward.
By inserting a gateway between the PSTN and the PBX, the customer benefits from retaining all their existing PBX functionality. The benefits for the user are significant: there's no re-learning how to pick up voice-mail or record an outgoing greeting, for example. Even fax and modem traffic will be routed successfully because the gateway supports T.38 fax and G.711 for routing traffic up to V.90 speeds. There's also no need to upgrade the internal LAN, since calls are only translated into IP when they reach the outside world. This minimizes the effect on performance and voice quality since voice traffic is not dependent on network load.
Calls from remote VoIP destinations are routed by the gateway to the PBX, and where legal, the gateway can use dialed number analysis to allow calls to be routed from VoIP to PSTN, saving the customer long distance, or perhaps international toll costs. The PBX, and the user, will not see a difference.
Future possibilities
At a later point, the company may wish to move to pure IP or IP to the desktop, for example when their PBX service contract expires or when they are conducting a routine LAN refresh. In this context the gateway remains relevant and a usable investment, translating calls between the PSTN and the internal IP network where necessary. Additional capacity and functionality can be added, often through a remote software upgrade, allowing the investment in the gateway to deliver future value.
VoIP gateways are incredibly versatile: they can be used in scenarios ranging from green-field sites to enabling home working, and from making TDM hardware VoIP-compatible to eliminating international call costs by using a remote gateway to trunk calls back to a central site. The story of VoIP in business is only just beginning.
About the author
Tim Burne is VegaStream's CEO. He has an Honours degree in Engineering from Brunel University. He can be reached at info@vegastream.com.
