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VOIP Seminar Report

VOIP Seminar Report


1 History
2 Functionality
3 Implementation
o 3.1 Reliability
o 3.2 Quality of service
o 3.3 Difficulty with sending faxes
o 3.4 Emergency calls
o 3.5 Integration into global telephone number system
o 3.6 Single point of calling
o 3.8 Security
o 3.9 Pre-Paid Phone Cards
o 3.10 Caller ID
o 3.11 VoIM
4 Adoption
o 4.1 Mass-market telephony
o 4.2 Corporate and telco use
o 4.3 Use in Amateur Radio
o 4.4 Click to call
5 Legal issues in different countries
o 5.1 IP telephony in Japan
5.1.1 Telephone number for IP telephony in Japan
6 Technical details
VOIP Seminar Report


If youve never heard of Internet Telephony, get ready to change the way you think about long-
distance phone calls. Internet Telephony, or Voice over Internet Protocol, is a method for taking
analog audio signals, like the kind you hear when you talk on the phone, and turning them into
digital data that can be transmitted over the Internet.
How is this useful? Internet Telephony can turn a standard Internet connection into a way to place
free phone calls. The practical upshot of this is that by using some of the free Internet Telephony
software that is available to make Internet phone calls, you are bypassing the phone company (and its
charges) entirely.

Internet Telephony is a revolutionary technology that has the potential to completely rework the
worlds phone systems. Internet Telephony providers like Vonage have already been around for a
little while and are growing steadily. Major carriers like AT&T are already setting up Internet
Telephony calling plans in several markets around the United States, and the FCC is looking
seriously at the potential ramifications of Internet Telephony service.
Above all else, Internet Telephony is basically a clever reinvention of the wheel. In this article,
well explore the principles behind Internet Telephony, its applications and the potential of this
emerging technology, which will more than likely one day replace the traditional phone system

The interesting thing about Internet Telephony is that there is not just one way to place a call.

There are three different flavors of Internet Telephony service in common use today:


The simplest and most common way is through the use of a device called an ATA (analog telephone
adaptor). The ATA allows you to connect a standard phone to your computer or your Internet
connection for use with Internet Telephony.
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The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and
converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T
CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box,
plug the cable from your phone that would normally go in the wall socket into the ATA, and youre
ready to make Internet Telephony calls. Some ATAs may ship with additional software that is loaded
onto the host computer to configure it; but in any case, it is a very straightforward setup.

IP Phones

These specialized phones look just like normal phones with a handset, cradle and buttons. But
instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector.
IP phones connect directly to your router and have all the hardware and software necessary right
onboard to handle the IP call. Wi-Fi phones allow subscribing callers to make Internet Telephony
calls from any Wi-Fi hot spot.


This is certainly the easiest way to use Internet Telephony. You dont even have to pay for long-
distance calls. There are several companies offering free or very low-cost software that you can use
for this type of Internet Telephony. All you need is the software, a microphone, speakers, a sound
card and an Internet connection, preferably a fast one like you would get through a cable or DSL
modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-
computer calls, no matter the distance.

If youre interested in trying Internet Telephony, then you should check out some of the free Internet
Telephony software available on the Internet. You should be able to download and set it up in about
three to five minutes. Get a friend to download the software, too, and you can start tinkering with
Internet Telephony to get a feel for how it works.
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An overview of how VoIP works

Voice over Internet Protocol (VoIP) is a protocol optimized for the transmission of voice through the
Internet or other packet switched networks. VoIP is often used abstractly to refer to the actual
transmission of voice (rather than the protocol implementing it). VoIP is also known as IP
Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband.
VoIP is pronounced voyp.

Companies providing VoIP service are commonly referred to as providers, and protocols which are
used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP
protocols. They may be viewed as commercial realizations of the experimental Network Voice
Protocol (1973) invented for the ARPANET providers. Some cost savings are due to utilizing a
single network to carry voice and data, especially where users have existing underutilized network
capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free,
while VoIP to public switched telephone networks, PSTN, may have a cost that is borne by the VoIP
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Voice over IP protocols carry telephony signals as digital audio, typically reduced in data rate using
speech data compression techniques, encapsulated in a data packet stream over IP.

There are two types of PSTN to VoIP services: Direct Inward Dialing (DID) and access numbers.
DID will connect the caller directly to the VoIP user while access numbers require the caller to
input the extension number of the VoIP user.

IP Phone

An IP phone uses Voice over IP technologies allowing telephone calls to be made over the internet
instead of the ordinary PSTN system. The phones use protocols such as Session Initiation Protocol,
Skinny Client Control Protocol or one of various proprietary protocols such as that used by Skype. IP
phones can be simple software-based Softphones or purpose-built hardware devices that appear
much like an ordinary telephone or cordless phone or an ATA (analog telephony adapter) which
allows to reuse ordinary PSTN phones. One of the primary motivations for implementing such a
system is the lower calling cost. When calling other IP phones over the internet one only pays for the
usually fixed cost internet bandwidth.

It may have many features an analog doesn't support, such as e-mail-like IDs for contacts that may be
easier to remember than names or phone numbers.

Elements of an IP phone

1. Hardware
2. DNS client
3. STUN client
4. DHCP client (not commonly used)
5. Signalling stack (SIP, H323, Skinny, Skype, or others)
6. RTP stack
7. User interface

Hardware of a stand alone IP phone
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Hardware-based IP phone

The overall hardware may look like telephone or mobile phone. An IP phone has the following
hardware components.

Speaker/ear phone and microphone

Key pad / touch pad to enter phone number and text (not used for ATAs).
Display hardware to feedback user input and show caller-id/messages (not used for ATAs).
General purpose processor (GPP) to process application messages.
A voice engine or a Digital signal processor to process RTP messages. Some IC
manufacturers provides GPP and DSP in single chip.
ADC and DAC converters: To convert voice to digital data and vice versa.
Ethernet or wireless network hardware to send and receive messages on data network.
Power source might be a battery or DC source. Some IP phones receive electricity from
Power over ethernet.

Other devices

There are several WiFi enabled mobile phones and PDAs that come pre-loaded with SIP clients or
are at least capable of running IP telephony clients. Some IP phones may also support PSTN phone
lines directly.

Analog telephony adapters

These are usually rectangular boxes that are connected to the internet or Local area network using an
Ethernet port and have sockets to connect one or more PSTN phones. Such devices are sent out to
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customers who sign up with various commercial VoIP providers allowing them to continue using
their existing PSTN based telephones.

Another type of gateway device acts as a simple GSM base station and regular mobile phones can
connect to this and make VoIP calls. While a license is required to run one of these in most countries
these can be useful on ships or remote areas where a low-powered gateway transmitting on unused
frequencies is likely to go unnoticed.

Common features of IP phones

Caller ID

Dialing using name/ID: This is different from dialing from your mobile call register as user
need not to save a number to sip phone.
Locally stored and network-based directories
Conference and multiparty call
Call park
Call transfer and call hold
Preserving user name/ number when choosing a different service provider (not widely
Applications like weather report, Attendance in school and offices, Live news etc.

Disadvantages of IP phones

Requires internet access to make calls outside the Local area network unless a compatible
local PBX is available to handle calls to and from outside lines.
Non-PoE IP Phones and the routers they connect through need to have their own power
supply unlike PSTN phones which are supplied with power from the telephone exchange.
IP networks are often more prone to outages and congestion than analogue phone networks.

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1876 Invention of the telephone

1915 Call across the continent
1973 ARPANET/Network Voice Protocol
1995 - Volcatec
1996 DSP
1997 VoIP introduced/global communications
2000 residential acceptance


When did VoIP begin?

This standard of communication dates as far back as Alexander Bell and his invention of the
telephone, utilizing the same basic purpose and design. With the notion that one person can talk to
another person far away using some kind of device, in 1876 this device was the telephone, but in
1996, it can be found on the Internet. The first telephone call from one end of the American
continent to the other was made 87 years ago, on January 25, 1915.

The inspiration for this technology is the Internet capability oh allowing one computer to talk to
another. In the past, with limited technology, communication was only possible if both parties had
the same kind of soundcard with the latest drivers installed; otherwise the result was more like a
Half-Duplex walkie-talkie quality.

Long ago

Worldwide communication first started out

POTS - Plain Old Telephone Systems allowed local area calling, but was only available to the elite,
since there was a huge cost involved, considering the equipment and line placement. The POTS
network grew, as did its popularity and necessity (for individuals and corporations alike)

PSTN - Public Switched Telephone Networks

The industry quickly evolved to include nationwide and eventually global connectivity through the
phone company.

1973 Voice over IP or VoIP Protocols are used to carry voice signals over the IP network, a
commercial realization of the experimental Network Voice Protocol invented for the ARPANET.
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1995 the first Internet Phone Software appeared - Vocaltec

Vocaltec released the first internet phone software called Internet Phone.
Hobbyists began to recognize the potential of sending voice data packets over the Internet
instead of communicating through standard telephone service
Designed to run on a home PC
Uutilized sound cards, microphones and speakers.
Allowed PC users to avoid long distance charges
The software used the H.323 protocol instead of the SIP protocol that is more prevalent

Contemporary VoIP

uses a standard telephone hooked up to an Internet connection

early efforts in the history of VoIP required both callers to have a computer equipped with the
same software, as well as a sound card and microphone.
early applications hadpoor sound quality and connectivity
but showed that VoIP technology was useful and promising, considered the Skype of the 90s.
A major drawback in 1995 was the lack of broadband availability. Also software used with modems
resulted in poor voice quality vs. normal telephone call. It was still a major milestone as it
represented the first ever IP Phone.

Voice over IP began as the result of work done by some hobbyists in Israel in 1995 when only PC-
to-PC communication was available. Later in 1995, Vocaltec, Inc. released Internet Phone Software.
This software was designed to run on a home PC (486/33 MHz) with sound cards, speakers,
microphone, and modem. The software compressed the voice signal, translated it into voice packets,
and shipped it out over the Internet. The technology worked as long as both the caller and the
receiver had the same equipment and software. Although the sound quality was nowhere near that of
conventional equipment at the time, this effort represented the first IP phone.

VoIP came into existence as a result of work done by a few hobbyists in Israel in the year 1995 when
only PC-to-PC communication was in vogue. Later on during 1995, Vocaltec, Inc. released Internet
Phone Software. This particular software was intended to run on a home PC (486/33 MHz) with:

sound cards
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The software was used to compress the voice signal, convert it into voice packets, and then finally to
ship it out over the Internet. This particular technology worked as long as both the caller and the
receiver had the same tools and software. However, the sound quality was not even close to that of
the standard equipment in use at that point of time. This attempt can be termed as the first IP phone
that came into existence.


Vocaltec one of the true pioneers of VoIP - Internet Phone product

It had initial success with Internet Phone, and had a successful IPO in 1996 and was perhaps the first
true VoIP software application. It helped lay the groundwork to make VoIP mainstream and was
the first VoIP product on the shelves of Compusa and other retail outlets.

In the old days of VoIP there were full-duplex issues and soundcard full-duplex driver issues. If you
didnt have the latest sound card driver, youd get a half-duplex CB/walkie-talkie type experience.
The Internet hadnt really taken off at that point in history. You had to download the latest sound
card driver to get full-duplex VoIP sound.

In 1996 they released and officially invented the protocol and today they are leading providers of the
latest VoIP solutions. The technology is still fairly new and history is being written right now.

Historically, VoIP software focused mainly on the DSP (Digital Signal Processors), primarily due to
the components high representation in the design of VoIP platforms. Not surprisingly, OEMs
centered their design decisions on which DSP they intended to use, with the standard considerations
of performance, size, and power dissipation following suit.

The VoIP software vendors responded in kind by supplying the necessary codecs and data packaging
components necessary to run on the DSP, however this bottom-up approach left manufacturers to
fend for themselves with the most critical design elements, including system management, signaling,
call control, gateway control, and control plane interface. Often, the integration of these disparate
components was quite a difficult process, requiring the stitching together of algorithms and protocols
from many different suppliers. Consequently, system efficiency was sub-optimal, and time to market
was painfully slow.


(VoIP evolved gradually over the next few years)

PC to phone service offered by small companies.
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Phone to phone service soon followed (by using a computer to establish the connection)
email, cellular (mobile), and the Internet becoming standards for global communications

By 1998, VoIP traffic had grown to represent approximately 1% of all voice traffic in the United
States. Entrepreneurs were jumping on the bandwagon and were creating devices which enabled PC-
to-phone and phone-to-phone communication. Networking manufacturers such as Cisco and Lucent
introduced equipment that could route and switch the VoIP traffic and as a result by the year 2000,
VoIP traffic accounted for more than 3% of all voice traffic.

By 1998 VOIP had reached some potential. A number of entrepreneurs started setting up gateways to
allow first PC-to-Phone and later Phone-to-Phone connections. Some of these entrepreneurs started
by providing customers a facility to make free phone calls using the regular phone. Every phone call
which the user made had an advertisement at the beginning and at the end of the call. This service
was only available to users in North America. This service allowed the users to make free long
distance calls. This free to the customer marketing model, was sponsored by various advertising
companies or agencies. These services often required the services of a PC to originate the call,
although the actual communication was from phone to phone. At this stage, VOIP traffic
represented rather less than 1% of voice traffic.

In 1998 three IP switch manufacturers introduced equipment capable of switching. At present, most
IP switching and routing equipment suppliers offer VOIP as either a standard or as an option on their
mid-range and up equipment.

Voice over Internet Protocol had made considerable progress by the year 1998. A number of
organizations began to set up gateways to allow first PC-to-Phone and later Phone-to-Phone
connections. A few of these organizations started by providing users a facility to make free phone
calls using the regular phone. Each phone call that the user made started with an advertisement and
also had one at the end of the call. This particular service was offered only to users in North
America. This allowed the users to make free long distance calls. A number of advertising
companies or agencies sponsored this free to the customer promotional model. These kinds of
services, time and again, require a PC to originate the call, even if the actual communication is from
phone to phone.

Three IP switch manufacturers launched equipment, during the year 1998, which was capable of
being used for switching.

late 1990s
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VoIP service relied on advertising sponsorship to subsidize costs, as opposed to charging customers
for calls.

The gradual introduction of broadband Ethernet service allowed for greater call clarity and reduced
latency, (calls still had static or there was difficulty making connections between the Internet and
PSTN (public telephone networks).

startup VoIP companies were able to offer free calling service to customers from special locations.

VoIP hardware less computer dependent (breakthrough in VoIP history)Cisco Systems and Nortel
(hardware manufacturers) started producing VoIP equipment that was capable of switching, therefore
functions that previously had to be handled by a computers CPU, such as switching a voice data
packet into something that could be read by the PSTN (and vice versa) could now be done by another

Since 2000

VoIP usage has expanded dramatically

several different technical standards for VoIP data packet transfer and switching - each is
supported by at least one major manufacturer
No clear winner has yet emerged to adopt the role of a universal standard.
Service has also been extended to residential users
While companies often switch to VoIP to save on both long distance and infrastructure costs,
VoIP has gone from being a fringe development to a mainstream alternative to standard
telephone service.
Currently, the majority of IP switching and routing equipment suppliers offer VoIP on their mid-
range and up equipment, either as standard equipment or as an option. Voice over Internet Protocol
traffic was in excess of 3% of voice traffic by the year 2000, and it is expected that it would grow
rapidly to somewhere between 25% and 40% of all international voice traffic by the year 2005.


Voice quality issues have long since been addressed and VoIP traffic can be prioritized over data
traffic to ensure reliable, clear sounding, unbroken telephone calls. Revenue from VoIP equipment
sales alone are projected to reach around $3 billion this year and are being forecast to be over $8.5
billion by the end of 2008. This is primarily being driven by low cost unlimited calling plans and the
abundance of enhanced and useful telephony features associated with VoIP technology.
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This is a phenomenal growth rate and with the rapid introduction of Video over IP fueling demand,
the future of this technology is truly exciting and will enable us to enjoy products that our
grandparents and even parents never thought were possible. Video over IP follows the same concept
as VoIP but in this case enables the transmission of video signals. As such, video phones are
becoming more common than you would think, and many companies are already offering attractive
packages. One of our featured partners, Packet8 already has a video phone offering.

Voice over Internet Protocol, VoIP or Broadband phone service as it is often referred to, is changing
the telephony world. Traditional phone lines are slowly being phased out as businesses and
households around the world embrace the benefits and features that VoIP technology has to offer.


VOIP has now become one of the most technologically advanced communications platform in the

Next 5 years
According to experts, with VoIPs increasing Quality of Service (QoS) and universality of added
features, it will occupy a major percentage of all communications


VoIP can facilitate tasks and provide services that may be more difficult to implement or expensive
using the more traditional PSTN. Examples include:

The ability to transmit more than one telephone call down the same broadband-connected
telephone line. This can make VoIP a simple way to add an extra telephone line to a home or
3-way calling, call forwarding, automatic redial, and caller ID; features that traditional
telecommunication companies (telcos) normally charge extra for.
Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.)
Most of the difficulties of creating a secure phone over traditional phone lines, like digitizing
and digital transmission are already in place with VoIP. It is only necessary to encrypt and
authenticate the existing data stream.
Location independence. Only an internet connection is needed to get a connection to a VoIP
provider. For instance, call center agents using VoIP phones can work from anywhere with a
sufficiently fast and stable Internet connection.
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Integration with other services available over the Internet, including video conversation,
message or data file exchange in parallel with the conversation, audio conferencing,
managing address books, and passing information about whether others (e.g. friends or
colleagues) are available online to interested parties.


Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential
order, or provide Quality of Service (known as QoS) guarantees, VoIP implementations face
problems dealing with latency and jitter. This is especially true when satellite circuits are involved,
due to long round trip propagation delay (400 milliseconds to 600 milliseconds for geostationary
satellite). The receiving node must restructure IP packets that may be out of order, delayed or
missing, while ensuring that the audio stream maintains a proper time consistency. This functionality
is usually accomplished by means of a jitter buffer in the voice engine.

Another challenge is routing VoIP traffic through firewalls and address translators. Private Session
Border Controllers are used along with firewalls to enable VoIP calls to and from a protected
enterprise network. Skype uses a proprietary protocol to route calls through other Skype peers on the
network, allowing it to traverse symmetric NATs and firewalls. Other methods to traverse firewalls
involve using protocols such as STUN or ICE.

VoIP challenges:

Available bandwidth
Delay/Network Latency
Packet loss
Pulse dialing to DTMF translation

Many VoIP providers do not translate pulse dialing from older phones to DTMF. The VoIP user may
use a VoIP Pulse to Tone Converter, if needed.[citation needed]
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Fixed delays cannot be controlled but some delays can be minimized by marking voice packets as
being delay-sensitive (see, for example, Diffserv).

The principal cause of packet loss is congestion, which can be controlled by congestion management
and avoidance. Carrier VoIP networks avoid congestion by means of teletraffic engineering.

Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a
jitter buffer upon arrival and before producing audio, although increases delay. This avoids a
condition known as buffer underrun, in which the voice engine is missing audio since the next voice
packet has not yet arrived.

Common causes of echo include impedance mismatches in analog circuitry, and acoustic coupling of
the transmit and receive signal at the receiving end.


Conventional phones are connected directly to telephone company phone lines, which in the event of
a power failure are kept functioning by back-up generators or batteries located at the telephone
exchange. However, household VoIP hardware uses broadband modems and other equipment
powered by household electricity, which may be subject to outages in the absence of a
uninterruptible power supply or generator. Early adopters of VoIP may also be users of other phone
equipment, such as PBX and cordless phone bases, that rely on power not provided by the telephone
company. Even with local power still available, the broadband carrier itself may experience outages
as well. While the PSTN has been matured over decades and is typically reliable, most broadband
networks are less than 10 years old, and even the best are still subject to intermittent outages.
Furthermore, consumer network technologies such as cable and DSL often are not subject to the
same restoration service levels as the PSTN or business technologies such as T-1 connection.

Quality of service

Some broadband connections may have less than desirable quality. Where IP packets are lost or
delayed at any point in the network between VoIP users, there will be a momentary drop-out of
voice. This is more noticeable in highly congested networks and/or where there are long distances
and/or interworking between end points. Technology has improved the reliability and voice quality
over time and will continue to improve VoIP performance as time goes on.

It has been suggested to rely on the packetized nature of media in VoIP communications and transmit
the stream of packets from the source phone to the destination phone simultaneously across different
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routes (multi-path routing). In such a way, temporary failures have less impact on the communication
quality. In capillary routing it has been suggested to use at the packet level Fountain codes or
particularly raptor codes for transmitting extra redundant packets making the communication more

A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These
include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323),
H.248.30 and MGCP extensions. The RFC3611 VoIP Metrics block is generated by an IP phone or
gateway during a live call and contains information on packet loss rate, packet discard rate (due to
jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end
system delay, signal / noise / echo level, MOS scores and R factors and configuration information
related to the jitter buffer.

RFC3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a
call, and an end of call message sent via SIP RTCP Summary Report or one of the other signaling
protocol extensions. RFC3611 VoIP metrics reports are intended to support real time feedback
related to QoS problems, the exchange of information between the endpoints for improved call
quality calculation and a variety of other applications.

Difficulty with sending faxes

What is a fax?

Internet fax uses the internet to receive and send faxes.

Traditional faxing involves sending a scanned copy of a document (a facsimile) from one fax
machine to another, over the phone network. Internet faxing (or "online faxing") is a general term
which can refer to one of several methods of achieving this over the Internet - with a goal of both
reduced costs and increased functionality over traditional faxing.

Depending on the specific method/implementation (see below), advantages of using the internet can

1. no extra telephone line required for the fax

2. paperless communication, integrated with email
3. send and receive multiple faxes simultaneously
4. reduction in phone costs
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Traditional fax

The traditional method for sending faxes over phone lines (PSTN)

Fax machine Phoneline Fax machine

A fax machine is an electronic instrument composed of a scanner, a modem, and a printer. It

transmits data in the form of pulses via a telephone line to a recipient, usually another fax machine,
which then transforms these impulses into images, and prints them on paper.

The traditional method requires a phone line, and only one fax can be connected to send or receive at
a time.

Computer-based faxing

As modems came into wider use with personal computers, the computer was used to send faxes
directly. Instead of first printing a hard copy to be then sent via fax machine, a document could now
be printed directly to the software fax, then sent via the computer's modem. Receiving faxes was
accomplished similarly.

Computer Phone line Fax machine

Fax Machine Phone line Computer

A disadvantage of receiving faxes this way is that the computer has to be turned on and running the
fax software to receive any faxes.
Note: This method is distinct from Internet faxing as the information is sent directly over the
telephone network, not over the Internet.

Internet fax servers/gateways

The Internet has enabled development of several other methods of sending and receiving a fax. The
more common method is an extension of computer-based faxing, and involves using a fax
server/gateway to the Internet to convert between faxes and emails. It is often referred to as "fax to
mail" or "mail to fax". This technology is more and more replacing the traditional fax machine
because it offers the advantage of dispensing with the machine as well as the additional telephone

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Fax machine Phone line Fax gateway email message (over Internet)
computer email account

A fax is sent via the Public Switched Telephone Network (PSTN) on the fax server, which receives
the fax and converts it into PDF or TIFF format, according to the instructions of the user. The fax is
then transmitted to the Web server which posts it in the Web interface on the account of the
subscriber, who is alerted of the reception by an email containing the fax in an attached file and
sometimes by a message on his mobile phone.


Computer Internet Fax gateway Phone line Fax machine

From his/her computer, in the supplier Web site, the user chooses the document s/he wants to send
and the fax number of the recipient. When sending, the document is usually converted to PDF format
and sent by the Web server to the fax server, which then transmits it to the recipient fax machine via
the Standard Telephone Network. Then the user receives a confirmation that the sending was carried
out, in his/her web interface and/or by email.

An Internet fax service allows one to send faxes from a computer via an Internet connection, thanks
to a Web interface usually available on the supplier's Web site. This technology has many

No fax machine no maintenance, no paper, toner expenditure, possible repairs, etc.

Mobility All actions are done on the Web interface; the service is thus available from any
computer connected to Internet, everywhere in the world.
Confidentiality The faxes are received directly on the account of the user; he is the only
one who can access it. The received faxes are not likely to be lost any more or read by the
wrong people.
No installation of software or hardware All actions are done on the Web interface of the
supplier, on the account of the user.
No telephone subscription for an additional line dedicated to the fax is required.
Many faxes can be sent or received simultaneously, and faxes can be received while the
computer is switched off.

Fax using Voice over IP
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Making phone calls over the Internet (Voice over Internet Protocol, or VoIP) has become
increasingly popular. Compressing fax signals is different from compressing voice signals, so a new
standard (T.38) has been created for this. If the VoIP adapter and gateway are T.38 compliant, most
fax machines can simply be plugged into the VoIP adapter instead of a regular phone line.

Fax machine VoIP adapter VoIP gateway Phone line Fax machine (or vice

As with regular faxes, only one fax can be sent or received at a time.

Fax using email

While the needs of computer-to-fax communications are well covered, the simplicity of quickly
faxing a handwritten document combined with the advantages of email are not.

"iFax" (T.37) was designed for fax machines to directly communicate via email. Faxes are sent as e-
mail attachments in a TIFF-F format.

iFax machine email message (over Internet) computer email account

iFax machine email message (over Internet) iFax machine (using email address)

A new fax machine (supporting iFax/T.37) is required, as well as a known email address for the
sending and receiving machines. This has limited the standard's use, though a system for looking up
a fax's email address based on its phone number is under development [1].

To work with existing fax machines, all iFax machines support standard faxing (requiring a regular
phone line). Alternatively, an iFax can be used in conjunction with a fax gateway.

iFax machine email message (over Internet) Fax gateway Phone line traditional
Fax machine (or vice versa)

The support of sending faxes over VoIP is still limited. The existing voice codecs are not designed
for fax transmission. An effort is underway to remedy this by defining an alternate IP-based solution
for delivering Fax-over-IP, namely the T.38 protocol. Another possible solution to overcome the
drawback is to treat the fax system as a message switching system, which does not need real time
data transmission - such as sending a fax as an email attachment (see Fax) or remote printout (see
Internet Printing Protocol). The end system can completely buffer the incoming fax data before
displaying or printing the fax image.
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Emergency calls

The nature of IP makes it difficult to locate network users geographically. Emergency calls,
therefore, cannot easily be routed to a nearby call center, and are impossible on some VoIP systems.
Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended
department. In the US, at least one major police department has strongly objected to this practice as
potentially endangering the public.[4]

Moreover, in the event that the caller is unable to give an address, emergency services may be unable
to locate them in any other way. Following the lead of mobile phone operators, several VoIP carriers
are already implementing a technical work-around.[citation needed] For instance, one large VoIP carrier
requires the registration of the physical address where the VoIP line will be used. When you dial the
emergency number for your country, they will route it to the appropriate local system. They also
maintain their own emergency call center that will take non-routable emergency calls (made, for
example, from a software based service that is not tied to any particular physical location) and then
will manually route your call once learning your physical location. [citation needed]

e911 is another method by which VOIP providers in the US are able to support emergency services.
The e911 emergency-calling system automatically associates a physical address with the calling
partys telephone number as required by the Wireless Communications and Public Safety Act of
1999 and is being successfully used by many VOIP providers to provide physical address
information to emergency service operators.

Integration into global telephone number system

While the traditional Plain Old Telephone Service (POTS) and mobile phone networks share a
common global standard (E.164) which allocates and identifies any specific telephone line, there is
no widely adopted similar standard for VoIP networks. Some allocate an E.164 number which can be
used for VoIP as well as incoming/external calls. However, there are often different, incompatible
schemes when calling between VoIP providers which use provider specific short codes.

Single point of calling

With hardware VoIP solutions it is possible to connect the VoIP router into the existing central
phone box in the house and have VoIP at every phone already connected. Software based VoIP
services require the use of a computer, so they are limited to single point of calling, though telephone
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sets are now available, allowing them to be used without a PC. Some services provide the ability to
connect WiFi SIP phones so that service can be extended throughout the premises, and off-site to any
location with an open hotspot.[5] However, note that many hotspots require browser-based
authentication, which most SIP phones do not support.[6]

Mobile phones & Hand held Devices

Telcos and consumers have invested billions of dollars in mobile phone equipment. In developed
countries, mobile phones have achieved nearly complete market penetration, and many people are
giving up landlines and using mobiles exclusively. Given this situation, it is not entirely clear
whether there would be a significant higher demand for VoIP among consumers until either public or
community wireless networks have similar geographical coverage to cellular networks (thereby
enabling mobile VoIP phones, so called WiFi phones or VoWLAN) or VoIP is implemented over 3G
networks. However, dual mode telephone sets, which allow for the seamless handover between a
cellular network and a WiFi network, are expected to help VoIP become more popular.[7]

Phones like the NEC N900iL, and later many of the Nokia Eseries and several WiFi enabled mobile
phones have SIP clients hardcoded into the firmware. Such clients operate independently of the
mobile phone network unless a network operator decides to remove the client in the firmware of a
heavily branded handset. Some operators such as Vodafone actively try to block VoIP traffic from
their network[8] and therefore most VoIP calls from such devices are done over WiFi.

Several WiFi only IP hardphones exist, most of them supporting either Skype or the SIP protocol.
These phones are intended as a replacement for PSTN based cordless phones but can be used
anywhere where WiFi internet access is available.

Another addition to hand held devices are ruggedized bar code type devices that are used in
warehouses and retail environments. These type of devices rely on inside the 4 walls type of VoIP
services that do not connect to the outside world and are solely to be used from employee to
employee communications.


Many consumer VoIP solutions do not support encryption yet, although having a secure phone is
much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to
eavesdrop on VoIP calls and even change their content. [9] There are several open source solutions
that facilitate sniffing of VoIP conversations. A modicum of security is afforded due to patented
audio codecs that are not easily available for open source applications, however such security
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through obscurity has not proven effective in the long run in other fields. Some vendors also use
compression to make eavesdropping more difficult. However, real security requires encryption and
cryptographic authentication which are not widely available at a consumer level. The existing secure
standard SRTP and the new ZRTP protocol is available on Analog Telephone Adapters(ATAs) as
well as various softphones. It is possible to use IPsec to secure P2P VoIP by using opportunistic
encryption. Skype does not use SRTP, but uses encryption which is transparent to the Skype

The Voice VPN solution provides secure voice for enterprise VoIP networks by applying IPSec
encryption to the digitized voice stream.

Pre-Paid Phone Cards

VoIP has become an important technology for phone services to travelers, migrant workers and
expatriates, who either, due to not having a fixed or mobile phone or high overseas roaming charges,
choose instead to use VoIP services to make their phone calls. Pre-paid phone cards can be used
either from a normal phone or from Internet cafes that have phone services. Developing countries
and areas with high tourist or immigrant communities generally have a higher uptake.

Caller ID

Caller ID support among VoIP providers varies, although the majority of VoIP providers now offer
full Caller ID with name on outgoing calls. When calling a traditional PSTN number from some
VoIP providers, Caller ID is not supported.

Caller ID (caller identification, CID, or more properly calling number identification) is a

telephone service that transmits a caller's number to the called party's telephone equipment during
the ringing signal, or when the call is being set up but before the call is answered. Where available,
caller ID can also provide a name associated with the calling telephone number. The information
made available to the called party may be made visible on a telephone's own display or on a separate
attached device.
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Caller ID is often helpful for tracing down prank calls and telemarketers. However, it can also
impede communication by enabling users to become evasive. The concept behind caller ID is the
value of informed consent; however, it also poses problems for personal privacy.

Caller ID is also known as calling line identification (CLI) when provided via an ISDN connection
to a PABX, while in some countries, the terms caller display, calling line identification
presentation, call capture, or just calling line identity are used; call display is the predominant
marketing name used in Canada (though customers often call it caller ID). CNID originated with
automatic number identification (ANI) in the United States

However, CNID and ANI are not the same thing. Caller ID is made up of two separate pieces of
information: the calling number and the billing (or subscriber) name where available. When an
originating phone switch sends out a phone number as caller ID, the telephone company receiving
the call is responsible for looking up the name of the subscriber in a database. Additionally, nothing
ensures that the number sent by a switch is the actual number where the call originated. It is very
easy for a telephone switch initiating the call to send any digit string desired as caller ID.

What is displayed as caller ID also depends on the equipment originating the call.

If the call originates on a plain old telephone service line (a standard loop start line) caller ID is
provided by the service providers local switch. Since the network does not connect the caller to the
callee until the phone is answered generally the caller ID signal cannot be altered by the caller. Most
service providers however, allow the caller to block caller ID presentation through a *XX feature

A call placed behind a private branch exchange (PBX) has more options. In the typical telephony
environment a PBX connects to the local service provider through PRI trunks. Generally, although
not absolutely, the service provider simply passes whatever calling line ID appears on those PRI
access trunks transparently across the PSTN [Public Service Telephone Network]. This opens up the
opportunity for the PBX administrator to program whatever number they choose in their external
phone number fields.

IP phone services like Vonage support PSTN gateway installations across North America and indeed
in a large number of locations across the world. These gateways egress calls to the local calling area,
thus avoiding long distance toll charges a key feature of the Vonage service. Vonage also allows a
local user to have a number located in a foreign exchange; the New York Caller could have a Los
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Angeles number. When that user places a call the calling line ID would be that of a Los Angeles
number although they are actually located in New York.

With Cell phones the biggest issue appears to be in the passing of calling line ID information through
the network. Cell phone companies must support interconnecting trunks to a significant number of
Wireline and PSTN access carriers. In order to save money it appears that many cell phone carriers
do not purchase the North American feature Group D or PRI trunks required to pass calling line ID
information across the network.

In the United States, caller ID information is sent to the called party by the telephone switch as an
analog data stream (similar to data passed between two modems), using Bell 202 modulation
between the first and second rings, while the telephone unit is still on hook. If the telephone call is
answered before the second ring, caller ID information will not be transmitted to the recipient. There
are two types of caller ID, number only and name+number. Number only caller ID is called Single
Data Message Format (SDMF), which provides the caller's telephone number, the date and time of
the call. Name+number caller ID is called Multiple Data Message Format (MDMF), which in
addition to the information provided by SDMF format, can also provide the directory listed name for
the particular number. Caller ID readers which are compatible with MDMF can also read the simpler
SDMF format, but an SDMF caller ID reader will not recognize an MDMF data stream, and will act
as if there is no caller ID information present, e.g. as if the line is not equipped for caller ID.


In North America, there is one code to disable caller ID. The code is *67. In the United Kingdom and
Ireland, 141 is the equivalent code. Australia uses 1831. New Zealand uses 0197, in NZ you can also
request Telecom to permanently disable your caller ID. Hong Kong uses 133. Israel uses *43. In
Denmark *31* is used to hide the CID, where callers has a permanent disable the same code is used
for revealing the number. Other countries and networks may vary. On GSM mobile networks, callers
may dial #31# [8] before the number they wish to call to disable it.


When the caller has caller ID disabled by default, there is a code to enable caller ID, which is
operator dependent. On GSM mobile networks, callers may dial *31# [9] before the number they wish
to call to enable caller ID.

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Voice over Instant Messaging (VoIM) presents VoIP as one communication mode among several,
with an IM user interface (contact list and presence) as the primary user experience. Many instant
messenger services added client-to-client or client-to-PSTN VoIP in the mid-2000s.

Instant Messaging (IM)

Is a form of real-time communication between two or more people based on typed text. The text is
conveyed via computers connected over a network such as the Internet.

Instant messaging (often abbreviated simply to IM) offers real-time communication and allows easy
collaboration, which might be considered more akin to genuine conversation than email's "letter"
format. In contrast to e-mail, the parties know whether the peer is available. Most systems allow the
user to set an online status or away message so peers are notified when the user is available, busy, or
away from the computer. On the other hand, people are not forced to reply immediately to incoming
messages. For this reason, some people consider communication via instant messaging to be less
intrusive than communication via phone. However, some systems allow the sending of messages to
people not currently logged on (offline messages), thus removing much of the difference between
Instant Messaging and email.

It is possible to save a conversation for later reference. Instant messages are typically logged in a
local message history which closes the gap to the persistent nature of e-mails and facilitates quick
exchange of information like URLs or document snippets (which can be unwieldy when
communicated via telephone).

Recently, many instant messaging services have begun to offer video conferencing features, Voice
Over IP (VoIP) and web conferencing services. Web conferencing services integrate both video
conferencing and instant messaging capabilities. Some newer instant messaging companies are
offering desktop sharing, IP radio, and IPTV to the voice and video features.

The term "instant messenger" is a service mark of Time Warner[3] and may not be used in software
not affiliated with AOL in the United States. For this reason, the instant messaging client formerly
known as Gaim or gaim announced in April 2007 that they would be renamed "Pidgin" [

Mobile Instant Messaging
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Mobile Instant Messaging (MIM) is a presence enabled messaging service that aims to transpose the
desktop messaging experience to the usage scenario of being on the move. While several of the core
ideas of the desktop experience on one hand apply to a connected mobile device, others do not: Users
usually only look at their phone's screen presence status changes might occur under different
circumstances as happens at the desktop, and several functional limits exist based on the fact that the
vast majority of mobile communication devices are chosen by their users to fit into the palm of their
hand. Some of the form factor and mobility related differences need to be taken into account in order
to create a really adequate, powerful and yet convenient mobile experience: radio bandwidth,
memory size, availability of media formats, keypad based input, screen output, CPU performance
and battery power are core issues that desktop device users and even nomadic users with connected n

Friend-to-friend networks

Instant Messaging may be done in a Friend-to-friend network, in which each node connects to the
friends on the friendslist. This allows for communication with friends of friends and for the building
of chatrooms for instant messages with all friends on that network.

Emotions are often expressed in shorthand. For example; lol. But a movement is currently underway
to be more accurate with the emotional expression. Real time reactions such as (chortle) (snort)
(guffaw) or (eye-roll) are rapidly taking the place of acronyms.

Business application

Instant messaging has proven to be similar to personal computers, e-mail, and the WWW, in that its
adoption for use as a business communications medium was driven primarily by individual
employees using consumer software at work, rather than by formal mandate or provisioning by
corporate information technology departments. Tens of millions of the consumer IM accounts in use
are being used for business purposes by employees of companies and other organizations.

In response to the demand for business-grade IM and the need to ensure security and legal
compliance, a new type of instant messaging, called "Enterprise Instant Messaging" ("EIM") was
created when Lotus Software launched IBM Lotus Sametime in 1998. Microsoft followed suit
shortly thereafter with Microsoft Exchange Instant Messaging, later created a new platform called
Microsoft Office Live Communications Server, and released Office Communications Server 2007 in
October 2007. Both IBM Lotus and Microsoft have introduced federation between their EIM systems
and some of the public IM networks so that employees may use a single interface to both their
internal EIM system and their contacts on AOL, MSN, and Yahoo!. Current leading EIM platforms
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include IBM Lotus Sametime, Microsoft Office Communications Server, and Jabber XCP. In
addition, industry-focused EIM platforms such as IMtrader from Pivot Incorporated, Reuters
Messaging, and Bloomberg Messaging provide enhanced IM capabilities to financial services

Comparison of VoIP software

Voice over IP (VoIP) software is used to conduct telephone-like voice conversations across IP based
networks. For residential markets, VOIP phone service is often cheaper than traditional PSTN phone
service and can remove geographic restrictions to telephone numbers (i.e. have a "New York" PSTN
phone number in Tokyo).

For enterprise or business markets, VoIP enables the enterprise to manage a single network (the IP
network) instead of separate voice and data networks, while enabling advanced and flexible
capabilities to the end user.

Calling ID is the identification of whom you are calling, or connecting to, as opposed to
caller ID identifying who calls you. Some Centrex telephone systems offer this feature.
Similarly, when one Skype user calls another Skype user, the caller can see the other party's
details and even an image or photograph they have chosen to represent their identity.
The inverse feature, giving the number originally dialed, is known as Direct Inward Dialing,
Direct Dialing Inward, or Dialed Number Identification System. This tells the PBX where to
route an incoming call, when there are more internal lines with external phone numbers than
there are actual incoming lines in a large company or other organisation.
List of telephony terminology
As a sidenote: Not all types of caller identification use 202-type modulation, nor do all
systems send the information between the first and second ring, e.g., British Telecom sends
the signal before the first ring, after a polarity reversal in the line. (Because of this most caller
ID software is not compatible with BT even if the Modem is) As a result, not all caller ID
devices are compatible from country to country or in the same country, even though the basic
phone system is the same. Some providers use FSK, others use the DTMF protocol.
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Mass-market telephony

A major development starting in 2004 has been the introduction of mass-market VoIP services over
broadband Internet access services, in which subscribers make and receive calls as they would over
the PSTN. Full phone service VoIP phone companies provide inbound and outbound calling with
Direct Inbound Dialing. Many offer unlimited calling to the U.S., and some to Canada or selected
countries in Europe or Asia as well, for a flat monthly fee.

These services take a wide variety of forms which can be more or less similar to traditional POTS.
At one extreme, an analog telephone adapter (ATA) may be connected to the broadband Internet
connection and an existing telephone jack in order to provide service nearly indistinguishable from
POTS on all the other jacks in the residence. This type of service, which is fixed to one location, is
generally offered by broadband Internet providers such as cable companies and telephone companies
as a cheaper flat-rate traditional phone service. Often the phrase VoIP is not used in selling these
services, but instead the industry has marketed the phrases Internet Phone, Digital Phone or
Softphone which is aimed at typical phone users who are not necessarily tech-savvy. Typically, the
provider touts the advantage of being able to keep ones existing phone number.

At the other extreme are services like Gizmo Project and Skype which rely on a software client on
the computer in order to place a call over the network, where one user ID can be used on many
different computers or in different locations on a laptop. In the middle lie services which also
provide a telephone adapter for connecting to the broadband connection similar to the services
offered by broadband providers (and in some cases also allow direct connections of SIP phones) but
which are aimed at a more tech-savvy user and allow portability from location to location. One
advantage of these two types of services is the ability to make and receive calls as one would at
home, anywhere in the world, at no extra cost. No additional charges are incurred, as call diversion
via the PSTN would, and the called party does not have to pay for the call. For example, if a
subscriber with a home phone number in the U.S. or Canada calls someone else within his local
calling area, it will be treated as a local call regardless of where that person is in the world. Often the
user may elect to use someone elses area code as his own to minimize phone costs to a frequently
called long-distance number.

For some users, the broadband phone complements, rather than replaces, a PSTN line, due to a
number of inconveniences compared to traditional services. VoIP requires a broadband Internet
connection and, if a telephone adapter is used, a power adapter is usually needed. In the case of a
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power failure, VoIP services will generally not function. Additionally, a call to the U.S. emergency
services number 9-1-1 may not automatically be routed to the nearest local emergency dispatch
center, and would be of no use for subscribers outside the U.S. This is potentially true for users who
select a number with an area code outside their area. Some VoIP providers offer users the ability to
register their address so that 9-1-1 services work as expected.

Another challenge for these services is the proper handling of outgoing calls from fax machines,
TiVo/ReplayTV boxes, satellite television receivers, alarm systems, conventional modems or
FAXmodems, and other similar devices that depend on access to a voice-grade telephone line for
some or all of their functionality. At present, these types of calls sometimes go through without any
problems, but in other cases they will not go through at all. And in some cases, this equipment can be
made to work over a VoIP connection if the sending speed can be changed to a lower bits per second
rate. If VoIP and cellular substitution becomes very popular, some ancillary equipment makers may
be forced to redesign equipment, because it would no longer be possible to assume a conventional
voice-grade telephone line would be available in almost all homes in North America and Western-
Europe. The TestYourVoIP website offers a free service to test the quality of or diagnose an Internet
connection by placing simulated VoIP calls from any Java-enabled Web browser, or from any phone
or VoIP device capable of calling the PSTN network.

Corporate and telco use

Although few office environments and even fewer homes use a pure VoIP infrastructure,
telecommunications providers routinely use IP telephony, often over a dedicated IP network, to
connect switching stations, converting voice signals to IP packets and back. The result is a data-
abstracted digital network which the provider can easily upgrade and use for multiple purposes.

Corporate customer telephone support often use IP telephony exclusively to take advantage of the
data abstraction. The benefit of using this technology is the need for only one class of circuit
connection and better bandwidth use. Companies can acquire their own gateways to eliminate third-
party costs, which is worthwhile in some situations.

VoIP is widely employed by carriers, especially for international telephone calls. It is commonly
used to route traffic starting and ending at conventional PSTN telephones.

Many telecommunications companies are looking at the IP Multimedia Subsystem (IMS) which will
merge Internet technologies with the mobile world, using a pure VoIP infrastructure. It will enable
them to upgrade their existing systems while embracing Internet technologies such as the Web,
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email, instant messaging, presence, and video conferencing. It will also allow existing VoIP systems
to interface with the conventional PSTN and mobile phones.

Electronic Numbering (ENUM) uses standard phone numbers (E.164), but allows connections
entirely over the Internet. If the other party uses ENUM, the only expense is the Internet connection.
Virtual PBX (or IP PBX) allow companies to control their internal phone network over an existing
LAN and server without needing to wire a separate telephone network. Users within this
environment can then use standard telephones coupled with an FXS, IP Phones connected to a data
port or a Softphone on their PC. Internal VoIP phone networks allow outbound and inbound calling
on standard PSTN lines through the use of FXO adapters.

Use in Amateur Radio

Sometimes called Radio Over Internet Protocol or RoIP, Amateur radio has adopted VoIP by linking
repeaters and users with Echolink, IRLP, D-STAR, Dingotel and EQSO. In fact, Echolink allows
users to connect to repeaters via their computer (over the Internet) rather than by using a radio. By
using VoIP Amateur Radio operators are able to create large repeater networks with repeaters all
over the world where operators can access the system with actual ham radios.

Ham Radio operators using radios are able to tune to repeaters with VoIP capabilities and use DTMF
signals to command the repeater to connect to various other repeaters, thus allowing them to talk to
people all around the world, even with line of sight VHF radios.

Click to call

Click-to-call is a service which lets users click a button and immediately speak with a customer
service representative. The call can either be carried over VoIP, or the customer may request an
immediate call back by entering their phone number. One significant benefit to click-to-call
providers is that it allows companies to monitor when online visitors change from the website to a
phone sales channel.

Click-to-call (CTC) refers to the process of converting web-based traffic into direct telephony
communication between an end user and some other entity. CTC processes vary depending upon
platforms, but there are two general styles of CTC. The first style uses the computer to complete the
call (typically PC-based VoIP.) Another style of CTC is the callback, where a user enters their phone
number and an intermediary service connects the end user to the merchant or other respective third
party. In this implmentation, these services tend to be more of an automatic dialing service than an
actual "click-to-call".
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When a user is browsing using a mobile phone, CTC features tend to be more literal. Phone numbers
are highlighted in the same manner a hyperlink would be. Clicking the phone number begins the
calling process.

One significant benefit of click-to-call providers is that it allows companies to monitor when online
visitors change from the website to a phone sales channel.

Legal issues in different countries

As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers,
governments are becoming more interested in regulating VoIP in a manner similar to PSTN
services,[10] especially with the encouragement of the state-mandated telephone
monopolies/oligopolies in a given country, who see this as a way to stifle the new competition.

In the U.S., the Federal Communications Commission now requires all interconnected VoIP service
providers to comply with requirements comparable to those for traditional telecommunications
service providers. VoIP operators in the U.S. are required to support local number portability; make
service accessible to people with disabilities; pay regulatory fees, universal service contributions, and
other mandated payments; and enable law enforcement authorities to conduct surveillance pursuant
to the Communications Assistance for Law Enforcement Act(CALEA). VoIP operators also must
provide Enhanced 911 service, disclose any limitations on their E-911 functionality to their
consumers, and obtain affirmative acknowledgements of these disclosures from all consumers. VoIP
operators also receive the benefit of certain U.S. telecommunications regulations, including an
entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via
wholesale carriers. Providers of nomadic VoIP servicethose who are unable to determine the
location of their usersare exempt from state telecommunications regulation.[11]

Some Latin American and Caribbean countries, fearful for their state owned telephone services, have
imposed restrictions on the use of VoIP, including in Panama where VoIP is taxed. In Ethiopia,
where the government is monopolizing telecommunication service, it is a criminal offense to offer
services using VoIP. The country has installed firewalls to prevent international calls being made
using VoIP. These measures were taken after a popularity in VoIP reduced the income generated by
the state owned telecommunication company.

In the European Union, the treatment of VoIP service providers is a decision for each Member
States national telecoms regulator, which must use competition law to define relevant national
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markets and then determine whether any service provider on those national markets has significant
market power (and so should be subject to certain obligations). A general distinction is usually
made between VoIP services that function over managed networks (via broadband connections) and
VoIP services that function over unmanaged networks (essentially, the Internet).

VoIP services that function over managed networks are often considered to be a viable substitute for
PSTN telephone services (despite the problems of power outages and lack of geographical
information); as a result, major operators that provide these services (in practice, incumbent
operators) may find themselves bound by obligations of price control or accounting separation.

VoIP services that function over unmanaged networks are often considered to be too poor in quality
to be a viable substitute for PSTN services; as a result, they may be provided without any specific
obligations, even if a service provider has significant market power.

The relevant EU Directive is not clearly drafted concerning obligations which can exist
independently of market power (e.g., the obligation to offer access to emergency calls), and it is
impossible to say definitively whether VoIP service providers of either type are bound by them. A
review of the EU Directive is under way and should be complete by 2007.

In India, it is legal to use VoIP, but it is illegal to have VoIP gateways inside India. This effectively
means that people who have PCs can use them to make a VoIP call to any number, but if the remote
side is a normal phone, the gateway that converts the VoIP call to a POTS call should not be inside

In the UAE, it is illegal to use any form of VoIP, to the extent that websites of Skype and Gizmo
Project dont work.

In the Republic of Korea, only providers registered with the government are authorized to offer VoIP
services. Unlike many VoIP providers, most of whom offer flat rates, Korean VoIP services are
generally metered and charged at rates similar to terrestrial calling. Foreign VoIP providers such as
Vonage encounter high barriers to government registration. This issue came to a head in 2006 when
internet service providers providing personal internet services by contract to United States Forces
Korea members residing on USFK bases threatened to block off access to VoIP services used by
USFK members of as an economical way to keep in contact with their families in the United States,
on the grounds that the service members VoIP providers were not registered. A compromise was
reached between USFK and Korean telecommunications officials in January 2007, wherein USFK
service members arriving in Korea before June 1, 2007 and subscribing to the ISP services provided
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on base may continue to use their U.S.-based VoIP subscription, but later arrivals must use a Korean-
based VoIP provider, which by contract will offer pricing similar to the flat rates offered by U.S.
VoIP providers.[12]

IP telephony in Japan

In Japan, IP telephony (IP IP Denwa ?) is regarded as a service applied by VoIP technology to

whole or a part of the telephone line. As of 2003, IP telephony services have been assigned telephone
numbers. IP telephony services also often include videophone/video conferencing services.
According to the Telecommunication Business Law, the service category for IP telephony also
implies the service provided via Internet, which is not assigned any telephone number. IP telephony
is basically regulated by Ministry of Internal Affairs and Communications (MIC) as a
telecommunication service. The operators have to disclose necessary information on its quality, etc.,
prior to making contracts with customers, and have an obligation to respond to their complaints
cordially. Many Japanese Internet service providers (ISP) are including IP telephony services. An
ISP who also provides IP telephony service is known as a ITSP (Internet Telephony Service
Provider). Recently, the competition among ITSPs has been activated, by option or set sales, in
connection with ADSL or FTTH services.

The tariff system normally applied to Japanese IP telephony is described below;

A call between IP telephony subscribers, limited to the same group, is usually free of charge.
A call from IP telephony subscribers to a fixed line or PHS is usually a uniformly fixed rate
all over the country.

Between ITSPs, the interconnection is mostly maintained at VoIP level.

Where the IP telephony is assigned normal telephone number (0AB-J), the condition for its
interconnection is considered same as normal telephony.
Where the IP telephony is assigned specific telephone number (050), the condition for its
interconnection is described below;
o Interconnection is sometimes charged. (Sometimes, its free of charge.) In case of
free-of-charge, mostly, communication traffic is exchanged via a P2P connection with
the same VoIP standard. Otherwise, certain conversions are needed at the point of the
VoIP gateway which incurs operating costs.
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Telephone number for IP telephony in Japan

Since September 2002, the MIC has assigned IP telephony telephone numbers on the condition that
the service falls into certain required categories of quality. Highly qualified IP telephony is assigned
a telephone number. Normally the number starts with 050. But, when its quality is so high that
customer almost could not tell the difference between it and a normal telephone and when the
provider relates its number with a location and provides the connection with emergency call
capabilities, the provider is allowed to assign a normal telephone number, which is a so-called 0AB-
J number.

Technical details

The two major competing standards for VoIP are the ITU standard H.323 and the IETF standard SIP.
Initially H.323 was the most popular protocol, though in the local loop it has since been surpassed
by SIP. This was primarily due to the latters better traversal of NAT and firewalls, although recent
changes introduced for H.323 have removed this advantage

However, in backbone voice networks where everything is under the control of the network operator
or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core
backbones[citation needed], and the vast majority of callers have little or no idea that their POTS calls are
being carried over VoIP.

Where VoIP travels through multiple providers softswitches the concepts of Full Media Proxy and
Signalling Proxy are important. In H.323, the data is made up of 3 streams of data: 1) H.225.0 Call
Signaling; 2) H.245; 3) Media. So if you are in London, your provider is in Australia, and you wish
to call America, then in full proxy mode all three streams will go half way around the world and the
delay (up to 500-600 ms) and packet loss will be high. However in signaling proxy mode where only
the signaling flows through the provider the delay will be reduced to a more user friendly 120-150

One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet
headers. Typically, to send a G.723.1 5.6 kbit/s compressed audio path requires 18 kbit/s of
bandwidth based on standard sampling rates. The difference between the 5.6 kbit/s and 18 kbit/s is
packet headers. There are a number of bandwidth optimization techniques used, such as silence
suppression and header compression. This can typically save 35% on bandwidth usage.
VOIP Seminar Report

VoIP trunking techniques such as TDMoIP can reduce bandwidth overhead even further by
multiplexing multiple conversations that are heading to the same destination and wrapping them up
inside the same packets. Because the packet header overhead is shared between many simultaneous
streams, TDMoIP can offer near toll quality audio with a per-stream packet header overhead of only
about 1 kbit/s.