For enterprises, leveraging VoIP means folding internal
voice communications onto the LAN/WAN data network thus
resulting in a single network to manage. Furthermore,
inasmuch as voice is turned into another applications
“running” on the data network, new productivity-enhancing
features can be devised. Also, with the use of a softphone
or a WiFi phone, incoming calls can be forwarded (through
the Internet) to the called party, wherever he is, as
long as he is connected to the Internet (mobility).
Because Internet voice is digital, it may offer features and
services that are not available with a traditional phone. If
you have a broadband internet connection, you need not maintain
and pay the additional cost for a line just to make telephone
With many Internet voice plans, onecan talk for as long as one
wants with any person in the world (the requirement is that
the other person has an Internet connection). You can also talk
with many people at the same time without any additional cost.
3. What are the benefits of VoIP and converged voice-data networks?
Economic: Decreased capital infrastructure and staff to support
a single network (e.g. one network of cabling, wiring closets),
Simplification of, and reduced cost of moves/adds/changes (MACs)
of personnel, reduced long distance/toll charges (especially
with multi-branch locations).
Increased Communication Functionality: More options available
(e.g. unified messaging), Cost to implement options are low
Increased Productivity: Scalability allows the network to grow
more easily, the flexibility allows for a dynamic changing user
4. Is there a difference between making a local call and a long distance call?
It also depends on the agreement you have with your VoIP service
provider. For instance: your company is located in San Francisco
and calls New York a lot. You can have a New York area code
and phone number which would incur no long distance charges
calling New York.
5. If I have an internet voice service, who can I call?
there is no limit to where you can call, just as a traditional
phone. Though some service providers may only allow calls to
other subscribers of their service. The call can be made to
a local number, a mobile phone, to a long distance number, or
an international number. You may even utilize the service to
speak with more than one person at a time. The person you are
calling does not need any special equipment, just a traditional
phone will do.
6. Can VoIP make and receive calls to/from PSTN lines?
A traditional phone system operating through an office PBX would
also be inoperative. In both cases, you require a power back-up
system, like a UPS (uninterrupted power supply) which would
provide power without any interruption when the power goes out.
8. Will faxing be affected by a VoIP installation?
You may still use your fax machine in the same way. Depending
on the agreement that you have with your service provider, you
might have many new fax features available through a software
application running on your desktop.
If your enterprise elects to use “soft phones”,
which is a software application running on your pc or laptop,
then your phone device would be replaced by the pc or laptop
and would need to be on to communicate through it.
13. Can I surf the web or send/receive email during VoIP calls?
VoIP allows users to surf the web or send/receive emails while
making and receiving VoIP calls. VoIP shares the broadband bandwidth
with other computers and devices on your local area network
(LAN) and prioritizes voice calls.
a small office environment (where there are up to 15 extensions),
a single high-speed DSL or cable modem connection to the Internet
should be fine. Approximately 800Kbps/3Mbps upload/download.
However, for larger offices, it is suggested that a T1 line
or better be used for the internet connection.
15. What type of service and equipment are needed for VoIP deployment?
Service Providers (VSPs) are the next generation telecom service
providers that provide interconnection between IP and PSTN networks.
They allow call origination and termination between these two
depends on your preference and budget. An ATA allows you to
use analog phones for VoIP. While this might save money, they
are missing some of the one touch feature keys (e.g. transfer,
hold, etc) and several other features.
IP-PBX (Private Branch Exchange) is a server-based application
providing, in a VoIP environment, the common telephony
features and services supported by traditional PBXs:
call transfers, conference calls, voicemail, music-on-hold,
auto call routing, and auto-attendant.
VoIP gateways are devices that take analog voice signals
and convert them to IP for transport over a LAN (Local
Area Network) or WAN (Wide Area Network).
Gateways are bridges between a VoIP network and the PSTN. While
a company would leverage its LAN/WAN network to support internal
voice communication with VoIP gateways are required to place/receive
calls to/from the PSTN.
term codec is a contraption of COder and DECoder. The
codec is where the analog to digital transformation
is performed. In the PSTN, the Pulse Code Modulation
(PCM) scheme is used to generate a 64kbps stream. More
“economical” coding schemes have been designed
to reduce this stream to 32, 16, and even 8 kbps. The
main ITU codec standards are G.711, G.722, G.726, G.728,
G.729, G.723. This reduction in the number of kbps requires
comes at the price of reduced quality of the voice signal,
thus a required trade-off between bandwidth and quality
22. What is the relationship between codec and VoIP?
order to transfer voice (analog signal) over IP (internet),
the voice message must first be digitized. Codec basically transfers
voice into data in order to travel through the public internet
network. Presently codec algorithms are designed to reduce the
size of their voice packets to increase speeds of transmissions
and reduce bandwidth loads on the network.
signaling protocols are used to begin and end calls. They carry
the required information to locate end users, and negotiate
assigned actions depending on the end user’s device capabilities.
The following are common VoIP signaling protocols: SIP (Session
Initiation Protocol), H.323, Cisco SCCP (Skinny Client Control
Protocol), IAX (Inter-Asterisk Exchange), and MGCP (Media Gateway
24. How much bandwidth does a voice call use on my IP network?
is related to the codec being used. The G.711 codec uses 64kbps
of bandwidth. The higher transmission rate in the G.711 codec
is a common misconception about VoIP. The G.729 Codec compresses
voice to 8Kbps, while the G723.1 codec has the ability to compress
voice at two separate rates i.e. 6.3kbps and 5.3kbps. It should
be noted that packet overhead (typically 46 octets) will increase
these respective bandwidths, so it is best to choose a compressed
voice Codec to minimize this effect. For example, the total
required bandwidth including the packet header is about 9.33kbps
for G723.1 (5.3kbps) and about 10.40kbps for G723.1 (6.3kbps).
is a complex protocol and does not go through a firewall
unless special software has been written for the firewall.
SIP is a simpler protocol and can be routed through
firewalls by opening specific port numbers in advance.
and FXO stand for Foreign Exchange Subscriber and Foreign Exchange
Office respectively. The FXS interface delivers POTS service
from the local phone company (from the Central Office). The
FXS interface provides dial tone, battery current, and ring
voltage to the subscriber device. The FXO interface receives
POTS service from the local phone company (from the switch at
the CO). The FXO interface provides on-hook and off-hook indication.
Federal Communications Commission (FCC) has worked to create
an environment to promote competition and innovation that will
benefit consumers. Historically, the FCC has not regulated the
internet or the services provided over it. The FCC decided on
February 12, 2004, that a voice over internet service will be
an unregulated information service. The FCC also started proceeds
to examine the FCC’s role in this new environment of increased
consumer choice. The FCC organized an Internet Policy Working
Group to identify, evaluate and address policy and issues that
will arise as telecommunications services move to Internet-based
platforms. For more information on the FCC visit www.fcc.gov/ipwg.
they do not but they have taken a preliminary position to be
finalized in the third to fourth quarter of 2005.
In April 2004, the CRTC launched a hearing process towards
a regulatory framework for VoIP communications in Canada.
In the abstract of its Telecom Public Notice 2004-2
, the CRTC states “The Commission is of the
preliminary view that voice communication services using
IP that utilize telephone numbers based on the North
American Numbering Plan and provide universal access
to and/or from the Public Switched Telephone Network
(PSTN) (referred to in this public notice as "VoIP"
services) have functional characteristics that are the
same as circuit-switched voice telecommunications services.
In the Commission's preliminary view, its existing regulatory
framework should apply to VoIP services, including its
determinations related to forbearance.” Currently,
the CRTC is expected to issue its ruling by mid-may
quality is most commonly rated through a voice quality metric
called the Mean Opinion Score (MOS) The MOS is a 5 point scale
where 5 represents excellent voice quality and 1 represents
bad voice quality. A score of 4.0 or greater is considered
suitable for a voice call. The name MOS comes from the fact
that voice quality is measured by using people listening to
sounds in headphones and rating the quality of these sounds
from 1 to 5.
Internet was not designed to carry voice traffic. Indeed, the
Internet was designed to carry traffic between computers (and
PCs). This data traffic has attributes that are opposite to
voice traffic. Data is mostly connectionless while voice is
connection-oriented; data traffic is insensitive to delay while
voice is very sensitive to delay as well as to delay variation
(jitter). When voice and data traffic share the same IP network,
control is exercised to ensure voice quality, and provide priority
to the voice packets.
Several QoS mechanisms have been designed to manage traffic
classes with different service requirements. These mechanisms
are used currently in enterprise networks as well as at the
boundary with the Internet.