Next Generation Networks (NGNs) connect voice and data communications over a common IP-based network. This allows phone calls to be made over internet connections using Voice-over-IP (VoIP) instead of traditional analog or ISDN connections. Migrating to an NGN requires replacing analog and ISDN phones with IP phones and connecting to the network with a single connection instead of separate lines. Companies must plan their migration to an NGN carefully by assessing their current infrastructure and creating a migration master plan.
The Codex of Business Writing Software for Real-World Solutions 2.pptx
Askozia NGN white paper - 2016, English
1. Next Generation Networks at a glance
Next Generation Networks (NGNs), or
All-IP, are the future of voice and data
communications. Both worlds are
connected in a common network.
Voice-over-IP (VoIP) connections are
the successors to ISDN and analog
connections. In other words, phone
calls are made through an IP-based
internet connection.
A separate telephone connection or
NTBAs are no longer needed. A NGN is
characterized by all services using
the same common infrastructure with
one standardized protocol. In the
future, IT and telephony are no longer
two separate areas, but share the
same hardware.
All important standards for Unified
Communications (UC) are defined by
means of the Internet protocol (IP).
Other protocols build on IP, such as
the Session Initiation Protocol (SIP) for
establishing a telephone connection.
For the infrastructure in your
company, this means that all phones
require only a single connection to the
network for using the services it
provides.
With IP-based connections, the
migration to VoIP offers great
opportunities for companies to make
processes much more efficient and
merge the different forms of
communication with each other.
ISDN becomes SIP trunks
SIP is the most widely used connection
protocol for IP telephony. SIP trunks
are usually offered by Internet service
providers (ISPs) and replace all types
of analog and ISDN connections.
Whereas the number of lines is limited
for traditional analog and ISDN,
usually any number of lines and phone
numbers can be booked from SIP trunk
providers. This is a purely virtual
service. The expansion of existing SIP
trunks is usually done with a single
click in the web interface of the ISP, or
a phone call. ISPs provide more flexible
terms than analog or ISDN providers.
Instead of two channels per ISDN BRI
line or 32 channels per ISDN PRI line,
any additional number of lines and
numbers can be booked usually
without signing long-lasting contracts.
Time is ticking for analog phone systems
How to optimally prepare your business for Voice-over-IP
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The migration from analog or ISDN to Voice-over-IP can lead to
significant changes in business communications. For private
customers, the migration is generally easy to implement. Many
companies, however, face far-reaching changes. Join us on our
journey into the future of IP communications.
2. Fig.: beroNet analog-SIP gateway
A typical business telecommunications
environment does not only consist of
the actual phone system and phones,
but many other components. Door
intercoms or alarm systems are often
also connected to the phone system
and in some cases even have their
own telephone line.
Migration master plan
As with all strategically important
projects, companies must first obtain
an overview and determine the state
of the existing telecommunications
infrastructure. In addition to the
telephone system, this includes the
telecommunications periphery that
has been mentioned in the previous
section. The duration of ongoing
maintenance and service contracts
also need to be considered.
If it has not been done yet, the
company should create a plan for the
further development of their
telecommunications infrastructure.
The strategic planning points to
immediate and intermediate-term
investment. Before the move to VoIP, a
fundamental decision needs to be
done: to either continue to use your
current PBX hardware and build a
bridge to the new technology, or to
migrate completely to NGNs instead.
If you would like to continue using your
current hardware, additional hardware
such as gateways or interface cards
can translate between your ISDN
phone system, analog devices, and
the new VoIP connections. This allows
you to use the devices of your current
telecommunications infrastructure
mostly unchanged. However, you get
the full NGN functionality range only
by completely migrating your
infrastructure and devices.
Even blocks of 1000 phone numbers,
which were so far an exclusive feature
of PRI lines, can be booked with SIP
trunks. For additional voice channels,
no physical connections are required.
Porting phone numbers of existing
analog or ISDN lines towards IP
telephony, is also not a problem.
Since the numbers are no longer
wired, they can be passed between
vendors with little effort. Many SIP
providers even offer this service for
free. It is also possible to register
foreign SIP trunks to offer local phone
numbers to customers abroad, or to
call abroad at a reduced rate.
LAN now also does telephony
VoIP uses the infrastructure of the
data network (LAN) to connect and
operate all devices, such as IP PBX
and IP phones. NTBAs and splitters
are eliminated, as well as separate
cabling for the telephone system.
Instead of expensive phones that are
compatible with only one type of PBX,
users can now choose between a wide
range of IP and soft phones thanks to
open standards. Devices and services
can be used flexibly from anywhere in
the network. In short, the user
receives more performance, whereas
costs are reduced.
No need to fear the change
Many companies still hesitate to
switch to VoIP. Some fear the high
workload, or do not want to replace
their current phone system that may
have been taken into operation just
recently. Others expect problems
with their telecommunications
infrastructure, as fax machines or
electronic cash terminals can not be
replaced overnight.
Fig.: Snom IP desk phone
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3. However, the local network (LAN)
must first meet some basic
requirements. All network devices
should support Quality of Service
(QoS) in order to transmit voice
packages prioritized and without
delay. The option to power your
devices via Power over Ethernet (PoE)
as well as setting up different virtual
networks (VLANs), is also useful and
leads to a leaner infrastructure.
Naturally, analog and ISDN phones
must be replaced by IP phones.
Planning stage
Generally, a deadline should be
defined for the migration towards
VoIP. As it may still require a longer
time frame, thorough planning and
pre-configuration are essential.
It is highly advisable to not rush into
dismantling your old phone system.
An example for the worst possible
case would be to instantly and
completely replace the old phone
system with the new IP PBX on a
Friday afternoon, and to directly start
into daily business on Monday
morning with the new technology,
thus risking total failure.
Besides all technical challenges, the
involvement and training of
employees is critical as well, in order
to gain their acceptance and
understanding. New technologies
inevitably lead to changes in business
processes.
For these reasons, a so-called soft
migration should be planned in
advance and eventually carried out,
in which both the old and new phone
systems coexist for a transitional
period.
Soft migration
In case of a soft migration, both
current and new PBX are operated in
parallel for a transitional period. This
allows to first test the functionality of
the new system in a small part of the
company, without affecting the daily
business telephony.
After the initial test and pilot stage, all
departments telecommunications are
gradually migrated. Each department
should be trained in the new PBX and
IP phones in order to use them
effectively.
At the end of the soft migration, all
proprietary system phones are
replaced by IP phones. The new IP PBX
is connected through a gateway to
your analog or ISDN connection, or
your old PBX. Further required analog
and ISDN devices such as fax
machines or alarm systems are also
connected through gateways to the
new PBX phone system.
Crash course IP PBX
Modern IP phone systems are
implemented in software practically
without exception. When talking about
modern IP phone systems, most
manufacturers speak of PBX (Private
Branch Exchange) or IP PBX. Since IP
phones, SIP trunks and other services
are registered at the PBX through the
local network (LAN) or the Internet, the
location of the PBX is initially
irrelevant.
An IP PBX running on a server in a data
center, is called cloud or hosted PBX.
Alternatively, an IP PBX can be
operated locally within the company
as a so-called in-house PBX. Both
solutions have individual advantages
and disadvantages. However, the
choice between a hosted or in-house
PBX should be made at the very
beginning if you plan to acquire a new
phone system, as it has far-reaching
consequences.
Hosted/Cloud PBX
In this case, the PBX software is
installed on an external server and
only accessible via internet.
Depending on the provider, you have
different access and configuration
rights. The pricing models usually are
user-based and differ greatly. Also, the
cloud PBX provider usually takes care
of the maintenance, and in many
cases a particular SIP trunk is
predetermined.
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4. The main advantage of hosted PBX is
the central location of the IP PBX. IP
phones can be registered to the PBX
independent of their location. This
way, employees in home offices or in
branch offices around the world, can
be easily integrated.
The initial costs for hosted PBX are
often lower as less own hardware is
required on site, and the costs are
incurred on a monthly base.
However, this may become more
expensive in the long term.
Hosted solutions also provide less
control as the PBX is located outside
the company and internet service
providers (ISPs) often can not be
freely selected. In addition, even
internal calls between employees are
routed through the Internet, and
therefore require a fast internet
connection.
In-House/On-Premise PBX
If you prefer full control over your IP
PBX and free choice of your ISP, in-
house or on-premise solutions are the
best choice. In this case, the PBX is
located on site, as you are used to
from your traditional phone system.
The initial costs of an in-house PBX are
higher, but the monthly costs are
much lower in return. Also SIP trunks
can be booked from any provider,
which is often not possible for hosted
solutions. Internal calls are routed via
the local network and therefore do not
require any internet bandwidth.
Virtual or Server Hardware
In-house solutions distinguish between
virtual solutions and PBX with own,
dedicated server hardware. A virtual
PBX can be set up easily on a server
within a virtual environment, such as
VirtualBox, VMware or HyperV.
This way, no additional hardware is
required and costs are lowered.
However, a failure of the server
implies also a failure of the telephone
system. Here, dedicated PBX hardware
is beneficial. The software runs on a
dedicated server, that generally has
been optimized for operation as a
telephone system.
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Askozia IP phone systems
Software PBX with an incredibly fair pricing
AskoziaPBX is an innovative IP PBX phone system, that was
specifically designed for the needs of small and medium-sized
businesses (SMEs). It provides professional telephony utilizing all
modern communication technologies.
Fig.: Snom IP desk phone and Askozia web interface
5. The AskoziaPBX software is the core
of the Askozia product family. Askozia
supports VoIP, as well as analog and
ISDN connections and devices.
The intuitive web interface allows you
to easily set up and manage the
whole system, and match it with your
specific business requirements.
User-friendly pricing
Askozia comes without any hidden
costs and does not limit your
telephony experience by additional
license fees per phone, user or
channel.
Software add-ons for AskoziaPBX
AskoziaPBX is built modularly. The PBX
can be extended by means of optional
software modules, to provide further
telephony features. These additional
software add-ons are simply activated
with license keys.
The Call Flow Editor is Askozia's most
popular add-on and allows to create
call flows via drag-and-drop. Even
complex interactive functions can be
implemented easily, such as custom
IVR menus (press 1 for ...) and ACD call
queues. This way, the Call Flow Editor
perfectly synchronizes the PBX with
your business processes.
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Fig.: Askozia Call Flow Editor
Askozia Hosted and Inhouse
Askozia supports both hosted and in-
house scenarios, and can be easily
integrated as a software PBX into
virtual environments such as
VirtualBox, VMware, or Hyper-V. For
in-house scenarios, full control of the
PBX and free choice of telephone
providers, Askozia offers complete
solutions including hardware.
Telephony Server
The Telephony Servers are Askozia's
complete solutions. AskoziaPBX is
pre-installed and the server works out
of the box.
Telephony Servers are available as
pure VoIP solutions or hybrid PBX with
additional analog or ISDN interfaces.
The systems are low-energy
consuming. Depending on the
equipment, the average consumption
is only 6-18 watts.
Partner network
Askozia has an international network
of resellers, hosting and installation
partners to support companies in
realizing telecommunication projects.
6. All solutions can be used on-site or in
the cloud, with IP phones of your
choice and can be configured and
managed through the most intuitive
user interface. No matter if you have
questions regarding the installation,
configuration or operation of
AskoziaPBX, our support team has
your back and offers you conditions
well-matched with your business
needs.
Learn more about how Askozia can
boost your business at askozia.com
About Askozia
Askozia started out in 2006 by
developing AskoziaPBX, a highly
intuitive telephone system firmware
for embedded appliances. Over the
years, Askozia has evolved into an
international developer of realtime IP
communication technologies and PBX
software for service providers, SMBs
and system integrators worldwide.
Askozia uses open standards for
interoperability and no proprietary
lock-in. The pricing is fair and
excludes licenses limiting the
numbers of users, phones or phone
lines.
Page 6
Fig.: Telephony Server with VoIP and analog ports