VOICE OVER INTERNET PROTOCOL
CONTENTS
¨ INTRODUCTION
¨ PROTOCOL
¨ ADOPTION
¨ ADVANTAGES
¨ CHALLENGES
¨ LEGAL ISSUES
¨ CONCLUSION
¨ REFERENCES
INTRODUCTION
Voice over Internet Protocol (VoIP) is a form of communication
that allows you to make phone calls over a broadband internet connection
instead of typical analog telephone lines. Basic VoIP access usually allows you
to call others who are also receiving calls over the internet. Interconnected
VoIP services also allow you to make and receive calls to and from traditional
landline numbers, usually for a service fee. Some VoIP services require a
computer or a dedicated VoIP phone, while others allow you to use your landline
phone to place VoIP calls through a special adapter.
VoIP
is becoming an attractive communications option for consumers. Given the trend
towards lower fees for basic broadband service and the brisk adoption of even
faster internet offerings, VoIP usage should only gain popularity with time.
However, as VoIP usage increases, so will the potential threats to the typical
user. While VoIP vulnerabilities are typically similar to the ones users face
on the internet, new threats, scams, and attacks unique to IP telephony are now
emerging.
Voice over Internet Protocol commonly refers to the communication protocols, technologies,
methodologies, and transmission techniques involved in the delivery of voice communications and multimedia
sessions over Internet Protocol (IP) networks, such as the Internet. Other
terms commonly associated with VoIP are IP
telephony, Internet telephony,
voice over broadband (VoBB), broadband telephony, IP communications, and broadband phone. Internet telephony refers to
communications services —voice, fax, SMS,
and/or voice-messaging applications— that are transported via the Internet,
rather than the public switched telephone network
(PSTN). The steps involved in originating a VoIP telephone call are signaling
and media channel setup, digitization of the analog voice signal, encoding,
packetization, and transmission as Internet
Protocol (IP) packets over a packet-switched network.
Voice
over IP (VoIP) can
facilitate tasks and deliver services that might be cumbersome or costly to
implement when using traditional PSTN:
- More than one phone call can be
transmitted on the same broadband phone line. This way, voice over IP can
facilitate the addition of telephone lines to businesses.
- Features that are usually charged
extra by telecommunication companies, such as call forwarding, caller ID
or automatic redialing, are simple with voice over IP technology.
- Unified
communications are secured with voice over IP technology, as it allows
integration with other services available on the internet such as video
conversation, messaging, etc.
PROTOCOL
- H.323
- Media Gateway Control
Protocol (MGCP)
- Session Initiation Protocol (SIP)
- Real-time Transport Protocol (RTP)
- Session Description Protocol
(SDP)
- Inter-Asterisk exchange (IAX)
The H.323 protocol
was one of the first VoIP protocols that found widespread implementation for
long-distance traffic, as well as local area network services. However, since the
development of newer, less complex protocols such as MGCP and SIP, H.323
deployments are increasingly limited to carrying existing long-haul network
traffic. In particular, the Session Initiation Protocol (SIP) has gained
widespread VoIP market penetration.
A
notable proprietary implementation is the Skype
protocol, which is in part based on the principles of peer-to-peer
(P2P) networking.
ADOPTION
Consumer market
Example of residential network including VoIP
A major development that started in 2004 was the introduction of
mass-market VoIP services that utilize existing broadband Internet access, by which
subscribers place and receive telephone calls in much the same manner as they
would via the public switched telephone network
(PSTN). Full-service VoIP phone companies provide inbound and outbound service
with Direct Inbound Dialing. Many offer unlimited
domestic calling for a flat monthly subscription fee. This sometimes includes
international calls to certain countries. Phone calls between subscribers of
the same provider are usually free when flat-fee service is not available A VoIP phone
is necessary to connect to a VoIP service provider. This can be implemented in
several ways:
- Dedicated
VoIP phones connect directly to the IP network using technologies such as
wired Ethernet
or wireless Wi-Fi.
They are typically designed in the style of traditional digital business
telephones.
- An analog telephone adapter is a device
that connects to the network and implements the electronics and firmware
to operate a conventional analog telephone attached through a modular
phone jack. Some residential Internet gateways and cable
modems have this function built in.
- A soft phone
is application software installed on a networked computer
that is equipped with a microphone and speaker, or headset. The
application typically presents a dial pad and display field to the user to
operate the application by mouse clicks or keyboard input.
PSTN and mobile
network providers
It is becoming increasingly common for telecommunications providers to
use VoIP telephony over dedicated and public IP networks to connect switching centers
and to interconnect with other telephony network providers; this is often
referred to as "IP backhaul".Smartphones
and Wi-Fi enabled
mobile phones may have SIP clients built into the firmware or available as an
application download.
Corporate use
Because of the bandwidth efficiency and low costs that VoIP technology
can provide, businesses are migrating from traditional copper-wire telephone
systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of
all new PBX lines
installed internationally were VoIP. VoIP solutions aimed at businesses have
evolved into unified communications services that treat
all communications—phone calls, faxes, voice mail, e-mail, Web conferences and
more—as discrete units that can all be delivered via any means and to any
handset, including cell phones. Two kinds of competitors are competing in this
space: one set is focused on VoIP for medium to large enterprises, while
another is targeting the small-to-medium business (SMB) market. VoIP allows
both voice and data communications to be run over a single network, which can
significantly reduce infrastructure costs.
ADVANTAGES
There are several advantages
to using Voice over IP. The biggest single advantage VoIP has over standard
telephone systems is cost. In addition, international calls using VoIP are
usually very inexpensive. One other advantage, which will become much more
pronounced as VoIP use climbs, calls between VoIP users are usually free. Using
services such as TrueVoIP, subscribers can call one another at no cost to
either party.
v Operational cost
·
VoIP
can be a benefit for reducing communication and infrastructure costs. Examples
include:
·
Routing
phone calls over existing data networks to avoid the need for separate voice
and data networks.
·
The
ability to transmit more than one telephone call over a single broadband
connection.
·
Secure
calls using standardized protocols (such as Secure Real-time Transport Protocol).
Most of the difficulties of creating a secure
telephone connection over traditional phone lines, such as digitizing and
digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate
the existing data stream.
v Portability
CHALLENGES
Quality of service
Communication on
the IP network is inherently less reliable in contrast to the circuit-switched
public telephone network, as it does not provide a network-based mechanism to
ensure that data packets are not lost, and are delivered in sequential order.
It is a best-effort network without fundamental Quality of Service (QoS) guarantees. Therefore,
VoIP implementations may face problems mitigating latency and jitter. By default,
network routers handle traffic on a first-come, first-served basis. Network
routers on high volume traffic links may introduce latency that exceeds
permissible thresholds for VoIP. Fixed delays cannot be controlled, as they are
caused by the physical distance the packets travel; however, latency can be
minimized by marking voice packets as being delay-sensitive with methods such
as DiffServ.
Layer-2 quality of
service
A
number of protocols that deal with the data
link layer and physical layer include quality-of-service mechanisms
that can be used to ensure that applications like VoIP work well even in
congested scenarios. Some examples include:
- IEEE
802.11e is an approved amendment to the IEEE
802.11 standard that defines a set of quality-of-service enhancements
for wireless LAN applications through modifications to the Media Access Control (MAC) layer. The
standard is considered of critical importance for delay-sensitive
applications, such as voice over wireless IP.
- IEEE
802.1p defines 8 different classes of service (including one dedicated
to voice) for traffic on layer-2 wired Ethernet.
- The ITU-T G.hn standard,
which provides a way to create a high-speed (up to 1 gigabit per second) Local area network using existing home
wiring (power lines, phone lines and coaxial cables). G.hn provides QoS by means
of "Contention-Free Transmission Opportunities" (CFTXOPs) which
are allocated to flows (such as a VoIP call) which require QoS and which
have negotiated a "contract" with the network controllers.
Susceptibility to
power failure
Telephones for traditional residential analog service are usually
connected directly to telephone company phone lines
which provide direct current to power most basic analog handsets independently
of locally available power.
IP Phones and
VoIP telephone adapters connect to routers
or cable
modems which typically depend on the availability of mains
electricity or locally generated power.[20]
Some VoIP service providers use customer premises equipment (e.g., cable modems)
with battery-backed power supplies to assure uninterrupted service for up to
several hours in case of local power failures. Such battery-backed devices typically
are designed for use with analog handsets.
Some VoIP service providers implement services to route calls to other
telephone services of the subscriber, such a cellular phone, in the event that
the customer's network device is inaccessible to terminate the call.
The susceptibility of phone service to power failures is a common
problem even with traditional analog service in areas where many customers
purchase modern telephone units that operate with wireless handsets to a base
station, or that have other modern phone features, such as built-in voicemail
or phone book features.
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. Sometimes, VoIP systems may route
emergency calls to a non-emergency phone line at the intended department; in
the United States, at least one major police department has strongly objected
to this practice as potentially endangering the public. A fixed line phone has
a direct relationship between a telephone number and a physical location. If an
emergency call comes from that number, then the physical location is known.
Lack of redundancy
The historical separation of IP networks and the PSTN provided
redundancy when no portion of a call was routed over IP network. An IP network
outage would not necessarily mean that a voice communication outage would occur
simultaneously, allowing phone calls to be made during IP network outages. When
telephone service relies on IP network infrastructure such as the Internet, a
network failure can isolate users from all telephony communication, including Enhanced
911 and equivalent services in other locales.
Number portability
Local number portability (LNP) and Mobile number portability (MNP) also
impact VoIP business. In November 2007, the Federal Communications Commission
in the United States released an order extending number portability obligations
to interconnected VoIP providers and carriers that support VoIP providers.
Number portability is a service that allows a subscriber to select a new
telephone carrier without requiring a new number to be issued. Typically, it is
the responsibility of the former carrier to "map" the old number to
the undisclosed number assigned by the new carrier. This is achieved by
maintaining a database of numbers. A dialed number is initially received by the
original carrier and quickly rerouted to the new carrier. Multiple porting
references must be maintained even if the subscriber returns to the original
carrier. The FCC mandates carrier compliance with these consumer-protection
stipulations.
PSTN integration
E.164
is a global FGFnumbering standard for both the PSTN and PLMN. Most VoIP implementations support E.164 to allow calls
to be routed to and from VoIP subscribers and the PSTN/PLMN.VoIP
implementations can also allow other identification techniques to be used. For
example, Skype allows subscribers to choose "Skype names" (usernames)
whereas SIP implementations can use Uris similar to email
addresses. Often VoIP implementations employ methods of translating
non-E.164 identifiers to E.164 numbers and vice-versa, such as the Skype-In
service provided by Skype and the ENUM service in IMS and SIP.
Security
VoIP telephone systems are susceptible to attacks as are any
Internet-connected devices. This means that hackers who know about these
vulnerabilities (such as insecure passwords) can institute denial-of-service
attacks, harvest customer data, record conversations and break into voice
mailboxes. Another challenge is routing VoIP traffic through firewalls and network address translators. Private Session Border Controllers are used along
with firewalls to enable VoIP calls to and from protected networks. For
example, 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 NATs involve using protocols
such as STUN or Interactive Connectivity
Establishment (ICE).
Securing VoIP
To prevent the above security concerns government and military
organizations are using voice over secure IP (VoSIP), secure voice over IP
(SVoIP), and secure voice over secure IP (SVoSIP) to protect confidential and
classified VoIP communications. Secure voice over secure IP is accomplished by
encrypting VoIP with protocols such as SRTP or ZRTP. Secure voice over
IP is accomplished by using Type
1 encryption on a classified network, like SIPRNet. Public
Secure VoIP is also available with free GNU programs and in many popular
commercial VoIP programs via libraries such as ZRTP.
Caller ID
Caller ID
support among VoIP providers varies, but is provided by the majority of VoIP
providers.Many VoIP carriers allow callers to configure arbitrary caller ID
information, thus permitting spoofing
attacks. Business-grade VoIP equipment and software often makes it easy to
modify caller ID information, providing many businesses great flexibility. The Truth in Caller ID Act
became law in on December 22, 2010. This bill proposes to make it a crime in
the United States to "knowingly transmit misleading or inaccurate caller
identification information with the intent to defraud, cause harm, or
wrongfully obtain anything of value ..." Rules implementing the law
were adopted by the Federal Communications Commission
on June 20, 2011.
Compatibility with
traditional analog telephone sets
Some analog telephone adapters do not decode pulse dialing from older
phones. They may only work with push-button telephones using the touch-tone
system. The VoIP user may use a pulse-to-tone converter, if needed.
Fax handling
Support for sending faxes over VoIP implementations is still limited.
The existing voice codecs
are not designed for fax transmission; they are designed to digitize an analog
representation of a human voice efficiently. However, the inefficiency of
digitizing an analog representation (modem signal) of a digital representation
(a document image) of analog data (an original document) more than negates any
bandwidth advantage of VoIP. In other words, the fax "sounds" simply
do not fit in the VoIP channel. An alternative IP-based solution for delivering
fax-over-IP called T.38
is available. Sending faxes using VoIP is sometimes referred to as FoIP, or Fax
over IP.
Support for other
telephony devices
Another challenge for VoIP implementations is the proper handling of
outgoing calls from other telephony devices such as digital video recorders, satellite television receivers, alarm systems,
conventional modems
and other similar devices that depend on access to a PSTN telephone
line for some or all of their functionality.
These types of calls
sometimes complete without any problems, but in other cases they fail. 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 PSTN telephone line would
be available in consumer's homes.
LEGAL
ISSUES
As the popularity of VoIP grows, governments are becoming more
interested in regulating VoIP in a manner similar to PSTN services.
Another legal issue that the US Congress is debating
concerns changes to the Foreign Intelligence Surveillance Act. The issue in
question is calls between Americans and foreigners. The National Security
Agency (NSA) is not authorized to tap Americans' conversations without a
warrant—but the Internet, and specifically VoIP does not draw as clear a line
to the location of a caller or a call's recipient as the traditional phone
system does. As VoIP's low cost and flexibility convinces more and more
organizations to adopt the technology, the surveillance for law enforcement
agencies becomes more difficult. VoIP technology has also increased security
concerns because VoIP and similar technologies have made it more difficult for
the government to determine where a target is physically located when communications
are being intercepted, and that creates a whole set of new legal challenges.
In the US, 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 US 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). "Interconnected" 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 US telecommunications regulations, including an
entitlement to interconnection and exchange of traffic with incumbent local exchange carriers
via wholesale carriers. Providers of "nomadic" VoIP service—those who
are unable to determine the location of their users—are exempt from state
telecommunications regulation.
Historical
milestones
- 1973:
Network Voice Protocol (NVP) developed by Danny Cohen and others to carry
real time voice over Arpanet.
- 1974: The Institute of
Electrical and Electronic Engineers (IEEE) published a paper titled
"A Protocol for Packet Network Interconnection".
- 1974:
Network Voice Protocol (NVP) first tested over Arpanet in August 1974,
carrying 16k CVSD encoded voice - first implementation of Voice over IP
- 1977: Danny
Cohen, Vint Cert, Jon Postel agree to separate IP from TCP, and create UDP
for carrying real time traffic
- 1981: IPv4 is described
in RFC 791.
- 1985: The National Science Foundation
commissions the creation of NSFNET.
- 1986:
Proposals from various standards organizations for Voice over ATM,
in addition to commercial packet voice products from companies such as StrataCom
- 1992: Voice
over Frame Relay standards development within Frame Relay Forum
- 1994: First
Voice Over IP application (Freeware for Linux) [61]
- 1995: Vocal Tec
releases the first commercial Internet phone software.[62][63]
- 1996:
- ITU-T begins
development of standards for the transmission and signaling of voice
communications over Internet Protocol networks with the H.323 standard.
- US
telecommunication companies petition the US Congress to ban Internet
phone technology.
- 1997: Level 3 began development of its first softswitch,
a term they coined in 1998.
- 1999:
- The Session Initiation Protocol (SIP)
specification RFC 2543
is released.
- Mark Spencer of Digium
develops the first open source private branch exchange (PBX)
software (Asterisk).
- 2004:
Commercial VoIP service providers proliferate.
CONCLUSION
The speed of development and the
widespread impacts of VoIP will be comparable to that of the Internet. VoIP is
truly a disruptive technology that will pose many challenges to BT, its staff
and its unions. The year 2003 saw VoIP move from being a fringe interest of
high tech enthusiasts to a concept central to the planning of all major telecommunications
operators. We shall be hearing a great deal about VoIP in 2004 in the technical
media, in discussions with BT, and in discussions with members.
However,
in some ways, VoIP is the froth on the wave and the wave itself is the
21CN/NGN. VoIP is the most predictable growth feature of the new network, but
there will be many others. Here, in the UK, 2004 also sees a strategic review
of telecommunications by the new super regulator Ofcom - the most fundamental
examination of the regulatory structure since the duopoly review in the early
1980s. Consequently 2004 could well prove to be a watershed year for all
concerned with the British telecommunications industry.
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(5): 11–19.
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2011-06-21.
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