56516814 Seminar Report on Voip

8/2/2019 56516814 Seminar Report on Voip

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Definition

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 theworlds 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

entirely.

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:

ATA

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.

Computer-to-computer

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 InternetTelephony 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 aprotocol 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, Broadbandtelephony, Broadband Phone and Voice over Broadband.

VoIP is pronounced voyp.

Companies providing VoIP service are commonly referred to as providers, andprotocols which are

used to carry voice signals over the IP network are commonly referred to as Voice over IP orVoIP

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 VoIPuser.

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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 orLocal area networkusing 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 regularmobile 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

supported).

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 networkunless 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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Contents

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

HISTORY of VoIP

1876 Invention of the telephone

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1915 Call across the continent

1973 ARPANET/Network Voice Protocol

1995 - Volcatec

1996 DSP

1997 VoIP introduced/global communications

2000 residential acceptance

HISTORY

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

today.

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

speakers

microphone

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modem

The software was used to compress the voice signal, convert it into voice packets, and then finally toship 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.

1996

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.

1998

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(VoIP evolved gradually over the next few years)

PC to phone service offered by small companies.

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-Phoneconnections. 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.

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Three IP switch manufacturers launched equipment, during the year 1998, which was capable of

being used for switching.

late 1990s

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

device

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.

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.

2005

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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.

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.

2006

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

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

Functionality

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

office.

3-way calling, call forwarding, automatic redial, and caller ID; features that traditional

telecommunication companies (telcos) normally charge extra for.

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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.

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.

Implementation

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 orICE.

VoIP challenges:

Available bandwidth

Delay/Network Latency

Packet loss

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Jitter

Echo

Security

Reliability

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]

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 ofteletraffic engineering.

Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a

jitterbuffer upon arrival and before producing audio, although increases delay. This avoids a

condition known asbuffer 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.

Reliability

Conventional phones are connected directly to telephone companyphone 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 equipmentpowered 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.

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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

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 orparticularly raptor codes for transmitting extra redundant packets making the communication more

reliable.

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 callquality 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

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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

include

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

Traditional fax

The traditional method for sending faxes overphone 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

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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

line.

Reception:

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 orTIFF 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.

Sending:

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 serverto 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 carriedout, 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

advantages:

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.

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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

Making phone calls over the Internet (Voice over Internet Protocol, or VoIP) has becomeincreasingly 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

versa)

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].

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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 beforedisplaying or printing the fax image.

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 ofmobile 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.

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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

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 orVoWLAN) or VoIP is implemented over3G

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 theNokia 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 fromtheir network[8] and therefore most VoIP calls from such devices are done over WiFi.

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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.

Security

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

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

provider.

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 countriesand areas with high tourist or immigrant communities generally have a higher uptake.

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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.

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 ofinformed consent; however, it also poses problems for personalprivacy.

Caller ID is also known as calling line identification (CLI) when provided via an ISDN connectionto 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

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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

code.

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 indeedin 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

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 isanswered 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.

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Disabling

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.

Enabling

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.

VoIM

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 ofreal-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 oraway 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 viaphone. 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.

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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 markof 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 formerlyknown as Gaim or gaim announced in April 2007 that they would be renamed "Pidgin"[

Mobile Instant Messaging

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.

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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 Server2007 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

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

companies.

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).

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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, orDialed 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.

Mass-market telephony

A major development starting in 2004 has been the introduction ofmass-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 orAsia 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

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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 ofSIP 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

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

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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 lowerbits per second

rate. If VoIP and cellularsubstitution 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 dedic

56516814 Seminar Report on Voip

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