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VoIP Connection(5056 bytes)
August 1999

VoIP: Part II - The Technical Deployment


There are many components required for launching these IP telephony services, and related factors for consideration.

A VoIP gateway bridges calls between the PSTN and an IP network. Gateways support different types of interfaces — analog or digital. An analog interface connects to the PSTN with the same interface as a regular “black” phone. Analog interfaces are almost identical worldwide, and are probably the easiest way to connect a gateway to the PSTN. However, they are not a scalable solution, and do not have good support for some calling features, especially when it comes to call progress indications such as disconnect signals and caller ID.

Digital lines come in different configurations. Some are unique to specific countries, such as T1 (United States and other countries), and E1 (European and other countries). Digital lines also come in different “flavors,” including ISDN and E&M. Common digital lines used for VoIP equipment today can carry up to 30 phone lines per single interface. These lines contain digital signaling information that improves functionality and enables enhanced services. Because of the wide variety of digital configurations, carriers should make sure that equipment supports the specific line interface that exists where services are being deployed. Check with your local telephony service provider for the line type used in a location.

A service provider can start with one gateway, but should consider redundancy for uninterrupted service. Having two gateways each with half the capacity needed might be a little more expensive than a single full capacity gateway, but this configuration allows for continuous service in the unlikely event that one goes down or requires a technical fix or upgrade.

Gatekeepers work in conjunction with a network of gateways. While gateways transfer calls between the PSTN and IP networks, gatekeepers provide the “brains” for a VoIP network. Among other things, gatekeepers provide network security by preventing unauthorized usage, call routing tables, and an interface to billing systems. As with gateways, gatekeeper redundancy can prevent service downtime. As a first step, component “duplication,” such as hard disk raid array and backup power supplies within a single gatekeeper, are a good start for realizing sufficient redundancy.

Network Management Station
The network management station enables the control and monitoring of the whole network from changing call routing tables to monitoring calls, gateway, and gatekeeper activities. The network management station provides the operator’s view and control of the system, but it is not part of the critical path of calls. In this instance, downtime on the network management station doesn’t mean downtime on the service side... though it can curtail and prevent a lot of headaches in that regard.

Billing System
The billing system communicates with the gatekeeper. The gatekeeper sends the billing system information such as call detail records (CDRs) to track calls. The billing system provides the gatekeeper information such as user authorization to perform a call, and available balance for a specific user. Depending on the application needed, the billing system can produce bills, reports, or notification on a user’s low balance. The billing system should be installed on a reliable system with built-in redundancy, since it is a critical component in the system.

Telephone Lines
The type of telephone connections used should be determined before ordering system components, because the hardware for the interface must fit the actual phone line(s). To decide this, an ITSP should consider a few factors. To start, gauge the number of simultaneous calls the system will need to handle when the service is launched, as well as the growth in number of lines expected in the short term and long term. The price of both the lines and the system, which vary with different interfaces, should be calculated. If the system supports less than 12 simultaneous calls when installed, and no additional lines will be needed in the near future, an analog line interface should be considered. Though the initial investment will be higher, if you need to add lines in the short run, a digital line interface might be the better choice because it will be significantly cheaper to expand down the road.

Internet Connection
Choosing the right Internet connection requires knowledge of what bandwidth a conversation will typically require. In general, this number will be around 10–20 Kbps for each conversation. An equipment vendor should be able to provide precise figures for calculating bandwidth. Keep in mind it’s not necessarily a linear relation between the number of lines supported and the total bandwidth required — some statistics might factor in additional bandwidth necessary for line increases.

Other considerations include carefully selecting the service provider to maximize the quality of service (QoS). As for telephony lines, factor in both the initial system and the growth pattern to determine what Internet connection to use. An interface may be priced less than another, but can be more costly when it comes to acquiring more bandwidth. Somewhat higher priced interfaces with a better ability to handle growth potential could prove the better approach. In any case, a constant connection with a fixed IP address is a “must,” regardless of the connection type.

There is a distinct advantage in PC-to-phone connectivity — whereby the person originating the call uses a computer, and the receiving party is on a “traditional” phone. In this case, the service provider needs only termination point(s), while offering virtually unlimited geographic origination coverage. For the technical deployment of a PC-to-phone service, there is a need for the following elements in at least one central location: A minimum of one gateway and gatekeeper, a network management station and billing system, appropriate telephone lines (analog or digital interface), and a fixed IP address Internet connection.

To launch a call, a destination telephone number is entered into a special software “client,” which is launched from the user’s Internet-enabled PC. The software connects over the Internet to the service provider’s gatekeeper, gets authorization to use a gateway, calls that specific gateway, and the gateway calls the destination phone number and bridges the user’s PC sound card and destination caller’s phone through the Internet. Note that in this scenario the network operating center (NOC) should be at a location in which calls to the destination desired will bear the lowest cost possible.

The location where the gatekeeper gives authorization to use a specific gateway is key to the billing process. The gatekeeper checks to see that a specific user has a unique identification number (the same principal as a calling-card PIN, or personal identification number). If approved to use the system, the gatekeeper passes the authorization on, and logs the event. The gatekeeper also saves the call start and end time. By analyzing the information and having a per-minute rate for a given destination, a billing system can determine the specific charge for the call.

Billing for PC-to-phone calls can be done in a few ways. With the prepaid approach, a user purchases a fixed amount and the cost of each call is deducted with each use. This is a preferred method for many service providers, since it doesn’t involve collection issues. The billing system should track continuously what the current balance is and notify the user if approaching zero. This will allow for the purchase of additional call time, or, disconnect calls when the credit is exhausted.

In a postpaid approach, the user is billed after the fact. This is common when the charge for the call is part of a phone bill. Most often, the PC-to-phone call is used to either get discounts on long-distance calls from a local telephone service provider, or as a means to place calls when there’s only one line available, already being used for an Internet connection. The billing system logs the calls and consolidates the information for each user in the monthly bill.

The advantage of phone-to-phone service is that users on both ends utilize existing and familiar equipment — the traditional telephone. However, in order for them to realize the advantage, there’s a larger investment in infrastructure for the ITSP. For the technical deployment of a phone-to-phone service, there is a need for the following elements: A minimum of two gateways — one at each site; a minimum of one gatekeeper; a network management station and billing system; telephone lines; and fixed IP address Internet connections in both locations.

A call starts from a user on a regular phone. The system can be one-, two-, or three-stage dialing. With one-stage dialing, the call is routed automatically to the VoIP system, and it’s completely transparent to the end user that the call is routed over an IP network. In two-stage dialing, the user calls an access number, gets a voice prompt, then dials the destination number. The system then uses the caller ID to identify and charge the caller. In three-stage dialing, the user calls an access number, dials a PIN number, and then proceeds with the destination number. The appropriate method depends on the application and on the technical limitations of the network. Is it a calling card or a long-distance provider service? Is caller ID available for all users who will rely on the service? Questions such as these will influence stage-dialing selection.

Service providers should also keep in mind that they can expand their geographical footprint by connecting to a clearinghouse that provides instant global termination points. The technical requirements for this depend on the clearinghouse infrastructure itself. As for billing, methods are similar to PC to phone, with pre- and postpaid versions possible.

Internet-based call centers are gaining momentum. Soon, it will be a competitive necessity for any company running a call center to allow its customers to connect with them via VoIP with the click of a “button” on the Web site. These calls are very similar to PC-to-phone calls, but with some interesting added features. For instance, the right system can additionally enable the call center host to show Web page information to the customer. The call is originated from a Web page — through a fixed phone number — with the destination side being charged for the call as a “toll-free” number.

There are a lot of factors that go into designing a system that will work best in handling the rollout of a specific VoIP service. Still, greater than these considerations is the potential payoff for those who are now offering and diligently preparing for the mass adoption of IP communications-based services. 

Lior Haramaty is a co-founder of VocalTec Communications, and belongs to the original group that started the VoIP industry. Haramaty has dealt with passing audio over data networks since the late 80s; VocalTec started shipping VoIP products in the early 90s. Haramaty has a multidisciplinary background in the business, technology, and marketing fields, is a co-inventor on VoIP patents, and initiated and spearheaded standards activities in the industry. The goal of this column is to clearly explain issues related to Voice (and other media) over Internet Protocol (VoIP) to anyone, including the “acronym-impaired” person. Requests for future column subjects to [email protected] are welcomed.

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