VoIP: Part II - The Technical Deployment
BY LIOR HARAMATY
There are many components required for launching these IP telephony services, and
related factors for consideration.
THE VoIP GATEWAY
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
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 operators 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 doesnt mean downtime on the service
side... though it can curtail and prevent a lot of headaches in that regard.
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 users 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.
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.
Choosing the right Internet connection requires knowledge of what bandwidth a
conversation will typically require. In general, this number will be around 1020
Kbps for each conversation. An equipment vendor should be able to provide precise figures
for calculating bandwidth. Keep in mind its 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 users Internet-enabled PC. The
software connects over the Internet to the service providers gatekeeper, gets
authorization to use a gateway, calls that specific gateway, and the gateway calls the
destination phone number and bridges the users PC sound card and destination
callers 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 doesnt 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 theres 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, theres 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 its 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
EMERGING CALL CENTER SERVICES
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
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@example.com are welcomed.