oice over Internet Protocol (
VoIP) is a methodology and group of technologies for 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 service.
The term
Internet telephony specifically refers to the provisioning of communications services (voice,
fax, SMS, voice-messaging) over the public
Internet, rather than via the
public switched telephone network (PSTN). The steps and principles involved in originating VoIP telephone calls are similar to traditional digital
telephony
and involve signaling, channel setup, digitization of the analog voice
signals, and encoding. Instead of being transmitted over a
circuit-switched network, however, the digital information is
packetized, and transmission occurs as
Internet Protocol (IP) packets over a
packet-switched network. Such transmission entails careful considerations about resource management different from
time-division multiplexing (TDM) networks.
Early providers of voice over IP services offered business models and
technical solutions that mirrored the architecture of the legacy
telephone network. Second-generation providers, such as
Skype,
have built closed networks for private user bases, offering the benefit
of free calls and convenience while potentially charging for access to
other communication networks, such as the PSTN. This has limited the
freedom of users to mix-and-match third-party hardware and software.
Third-generation providers, such as
Google Talk, have adopted
[1] the concept of
federated VoIP
– which is a departure from the architecture of the legacy networks.
These solutions typically allow dynamic interconnection between users on
any two domains on the Internet when a user wishes to place a call.
VoIP systems employ session control and signaling protocols to
control the signaling, set-up, and tear-down of calls. They transport
audio streams over IP networks using special media delivery protocols
that encode voice, audio, video with
audio codecs, and video codecs as
Digital audio by
streaming media.
Various codecs exist that optimize the media stream based on
application requirements and network bandwidth; some implementations
rely on
narrowband and
compressed speech, while others support
high fidelity stereo codecs. Some popular codecs include
μ-law and
a-law versions of
G.711,
G.722, which is a high-fidelity codec marketed as HD Voice by
Polycom, a popular open source voice codec known as
iLBC, a codec that only uses 8 kbit/s each way called
G.729, and many others.
VoIP is available on many
smartphones, personal computers, and on Internet access devices. Calls and SMS text messages may be sent over
3G or
Wi-Fi.
[2]
Protocols
Voice over IP has been implemented in various ways using both
proprietary protocols and protocols based on
open standards. Examples of the VoIP protocols are:
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.
These protocols can be used by special-purpose software, such as
Jitsi, or integrated into a web page (
web-based VoIP), like
Google Talk.
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 cablemodems have this function built in.
- A softphone
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."
[3][4]
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.
[5]
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 cellphones. 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.
[6]
VoIP allows both voice and data communications to be run over a
single network, which can significantly reduce infrastructure costs.
[7]
The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may run on commodity hardware, such as
personal computers. Rather than closed architectures, these devices rely on standard interfaces.
[7]
VoIP devices have simple, intuitive user interfaces, so users can
often make simple system configuration changes. Dual-mode phones enable
users to continue their conversations as they move between an outside
cellular service and an internal
Wi-Fi
network, so that it is no longer necessary to carry both a desktop
phone and a cellphone. Maintenance becomes simpler as there are fewer
devices to oversee.
[7]
Skype,
which originally marketed itself as a service among friends, has begun
to cater to businesses, providing free-of-charge connections between any
users on the Skype network and connecting to and from ordinary
PSTN telephones for a charge.
[8]
In the United States the Social Security Administration (SSA) is
converting its field offices of 63,000 workers from traditional phone
installations to a VoIP infrastructure carried over its existing data
network.
[9][10]
Quality of service
Communication on the IP network is perceived as less reliable in
contrast to the circuit-switched public telephone network because it
does not provide a network-based mechanism to ensure that data packets
are not lost, and are delivered in sequential order.
[citation needed] It is a best-effort network without fundamental
Quality of Service (QoS) guarantees. Therefore, VoIP implementations may face problems with
latency, packet loss, and
jitter.
[11][12]
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.
[11]
VoIP endpoints usually have to wait for completion of transmission of
previous packets, before new data may be sent. Although it is possible
to preempt (abort) a less important packet in mid-transmission, this is
not commonly done, especially on high-speed links where transmission
times are short even for maximum-sized packets.
[13] An alternative to preemption on slower links, such as dialup and
digital subscriber line (DSL), is to reduce the maximum transmission time by reducing the
maximum transmission unit.
But every packet must contain protocol headers, so this increases
relative header overhead on every link traversed, not just the
bottleneck (usually Internet access) link.
[13]
DSL modems provide Ethernet (or Ethernet over
USB) connections to local equipment, but inside they are actually
Asynchronous Transfer Mode (ATM) modems. They use
ATM Adaptation Layer 5
(AAL5) to segment each Ethernet packet into a series of 53-byte ATM
cells for transmission, reassembling them back into Ethernet frames at
the receiving end. A
virtual circuit identifier (VCI) is part of the 5-byte header on every ATM cell, so the transmitter can
multiplex the active virtual circuits (VCs) in any arbitrary order. Cells from the
same VC are always sent sequentially.
However, a majority of DSL providers use only one VC for each
customer, even those with bundled VoIP service. Every Ethernet frame
must be completely transmitted before another can begin. If a second VC
were established, given high priority and reserved for VoIP, then a low
priority data packet could be suspended in mid-transmission and a VoIP
packet sent right away on the high priority VC. Then the link would pick
up the low priority VC where it left off. Because ATM links are
multiplexed on a cell-by-cell basis, a high priority packet would have
to wait at most 53 byte times to begin transmission. There would be no
need to reduce the interface MTU and accept the resulting increase in
higher layer protocol overhead, and no need to abort a low priority
packet and resend it later.
ATM has substantial header overhead: 5/53 = 9.4%, roughly twice the
total header overhead of a 1500 byte Ethernet frame. This "ATM tax" is
incurred by every DSL user whether or not they take advantage of
multiple virtual circuits - and few can.
[11]
ATM's potential for latency reduction is greatest on slow links,
because worst-case latency decreases with increasing link speed. A
full-size (1500 byte) Ethernet frame takes 94 ms to transmit at
128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link,
this latency is probably small enough to ensure good VoIP performance
without MTU reductions or multiple ATM VCs. The latest generations of
DSL,
VDSL and
VDSL2, carry Ethernet without intermediate ATM/AAL5 layers, and they generally support
IEEE 802.1p priority tagging so that VoIP can be queued ahead of less time-critical traffic.
[11]
Voice, and all other data, travels in packets over IP networks with
fixed maximum capacity. This system may be more prone to congestion
[citation needed] and
DoS attacks[14] than traditional
circuit switched
systems; a circuit switched system of insufficient capacity will refuse
new connections while carrying the remainder without impairment, while
the quality of real-time data such as telephone conversations on
packet-switched networks degrades dramatically.
[11]
Fixed delays cannot be controlled as they are caused by the physical
distance the packets travel. They are especially problematic when
satellite circuits are involved because of the long distance to a
geostationary satellite and back; delays of 400–600 ms are typical.
When the load on a link grows so quickly that its switches experience
queue overflows, congestion results and data packets are lost. This
signals a transport protocol like
TCP to reduce its transmission rate to alleviate the congestion. But VoIP usually uses
UDP not TCP because recovering from congestion through retransmission usually entails too much latency.
[11]
So QoS mechanisms can avoid the undesirable loss of VoIP packets by
immediately transmitting them ahead of any queued bulk traffic on the
same link, even when that bulk traffic queue is overflowing.
The receiver must resequence IP packets that arrive out of order and
recover gracefully when packets arrive too late or not at all.
Jitter
results from the rapid and random (i.e. unpredictable) changes in queue
lengths along a given Internet path due to competition from other users
for the same transmission links. VoIP receivers counter jitter by
storing incoming packets briefly in a "de-jitter" or "playout"
buffer, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the
voice engine to play it. The added delay is thus a compromise between excessive latency and excessive
dropout, i.e. momentary audio interruptions.
Although jitter is a random variable, it is the sum of several other
random variables that are at least somewhat independent: the individual
queuing delays of the routers along the Internet path in question. Thus
according to the
central limit theorem, we can model jitter as a
gaussian random variable.
This suggests continually estimating the mean delay and its standard
deviation and setting the playout delay so that only packets delayed
more than several standard deviations above the mean will arrive too
late to be useful. In practice, however, the variance in latency of many
Internet paths is dominated by a small number (often one) of relatively
slow and congested "bottleneck" links. Most Internet backbone links are
now so fast (e.g. 10 Gbit/s) that their delays are dominated by the
transmission medium (e.g. optical fiber) and the routers driving them do
not have enough buffering for queuing delays to be significant.
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).
[15] 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 reliable.
[citation needed]
A number of protocols have been defined to support the reporting of
quality of service (QoS) and
quality of experience (QoE) for VoIP calls. These include
RTCP Extended Report (
RFC 3611),
SIP RTCP Summary Reports, H.460.9 Annex B (for
H.323),
H.248.30 and
MGCP
extensions. The RFC 3611 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 (because of jitter), packet loss/discard burst
metrics (burst length/density, gap length/density), network delay, end
system delay, signal / noise / echo level,
Mean Opinion Scores (MOS) and R factors and configuration information related to the jitter buffer.
RFC 3611 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. RFC 3611 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.
Rural areas in particular are greatly hindered in their ability to
choose a VoIP system over PBX. This is generally down to the poor access
to superfast broadband in rural country areas. With the release of 4G
data, there is a potential for corporate users based outside of
populated areas to switch their internet connection to 4G data, which is
comparatively as fast as a regular superfast broadband connection. This
greatly enhances the overall quality and user experience of a VoIP
system in these areas. This method was already trialled in rural
Germany, surpassing all expectations.
[16]
Layer 2
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.
PSTN integration
The Media VoIP Gateway connects the digital media stream, so as to
complete creating the path for voice as well as data media. It includes
the interface for connecting the standard PSTN networks with the ATM and
Inter Protocol networks. The Ethernet interfaces are also included in
the modern systems, which are specially designed to link calls that are
passed via the VoIP.
[17]
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.
[18]
VoIP implementations can also allow other identification techniques to
be used. For example, Skype allows subscribers to choose "Skype names"
[19] (usernames) whereas SIP implementations can use
URIs[20] 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
[21] and the
ENUM service in IMS and SIP.
[22]
Echo can also be an issue for PSTN integration.
[23] Common causes of echo include
impedance mismatches in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end.
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.
[24]
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.
A voice call originating in the VoIP environment also faces
challenges to reach its destination if the number is routed to a mobile
phone number on a traditional mobile carrier. VoIP has been identified
in the past as a
Least Cost Routing
(LCR) system, which is based on checking the destination of each
telephone call as it is made, and then sending the call via the network
that will cost the customer the least.
[25] This rating is subject to some debate given the complexity of call routing created by number portability. With
GSM
number portability now in place, LCR providers can no longer rely on
using the network root prefix to determine how to route a call. Instead,
they must now determine the actual network of every number before
routing the call.
Therefore, VoIP solutions also need to handle MNP when routing a
voice call. In countries without a central database, like the UK, it
might be necessary to query the
GSM
network about which home network a mobile phone number belongs to. As
the popularity of VoIP increases in the enterprise markets because of
least cost routing options, it needs to provide a certain level of reliability when handling calls.
MNP checks are important to assure that this quality of service is
met. Handling MNP lookups before routing a call provides some assurance
that the voice call will actually work.
Emergency calls
A telephone connected to a
land line
has a direct relationship between a telephone number and a physical
location, which is maintained by the telephone company and available to
emergency responders via the national emergency response service centers
in form of emergency subscriber lists. When an emergency call is
received by a center the location is automatically determined from its
databases and displayed on the operator console.
In IP telephony, no such direct link between location and
communications end point exists. Even a provider having hardware
infrastructure, such as a DSL provider, may only know the approximate
location of the device, based on the IP address allocated to the network
router and the known service address. However, some ISPs do not track
the automatic assignment of IP addresses to customer equipment.
[26]
IP communication provides for device mobility. For example, a residential broadband connection may be used as a link to a
virtual private network
of a corporate entity, in which case the IP address being used for
customer communications may belong to the enterprise, not being the
network address of the residential ISP. Such
off-premise extensions
may appear as part of an upstream IP PBX. On mobile devices, e.g., a 3G
handset or USB wireless broadband adapter, the IP address has no
relationship with any physical location known to the telephony service
provider, since a mobile user could be anywhere in a region with network
coverage, even roaming via another cellular company.
At the VoIP level, a phone or gateway may identify itself with a
Session Initiation Protocol (SIP) registrar by its account credentials. In such cases, the
Internet telephony service provider
(ITSP) only knows that a particular user's equipment is active. Service
providers often provide emergency response services by agreement with
the user who registers a physical location and agrees that emergency
services are only provided to that address if an emergency number is
called from the IP device.
Such emergency services are provided by VoIP vendors in the United States by a system called
Enhanced 911
(E911), based on the Wireless Communications and Public Safety Act of
1999. The VoIP E911 emergency-calling system associates a physical
address with the calling party's telephone number. All VoIP providers
that provide access to the public switched telephone network are
required to implement E911,
[26]
a service for which the subscriber may be charged. However,
end-customer participation in E911 is not mandatory and customers may
opt-out of the service.
[26]
The VoIP E911 system is based on a static table lookup. Unlike in
cellular phones, where the location of an E911 call can be traced using
assisted GPS
or other methods, the VoIP E911 information is only accurate so long as
subscribers, who have the legal responsibility, are diligent in keeping
their emergency address information current.
Fax support
Support for fax has been problematic in many VoIP implementations, as most voice digitization and compression
codecs
are optimized for the representation of human voice and the proper
timing of the modem signals cannot be guaranteed in a packet-based,
connection-less network. 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.
[27]
The T.38 protocol is designed to compensate for the differences
between traditional packet-less communications over analog lines and
packet-based transmissions which are the basis for IP communications.
The fax machine could be a traditional fax machine connected to the
PSTN, or an ATA box (or similar). It could be a fax machine with an
RJ-45 connector plugged straight into an IP network, or it could be a
computer pretending to be a fax machine.
[28]
Originally, T.38 was designed to use UDP and TCP transmission methods
across an IP network. TCP is better suited for use between two IP
devices. However, older fax machines, connected to an analog system,
benefit from UDP near real-time characteristics due to the "no recovery
rule" when a UDP packet is lost or an error occurs during transmission.
[29]
UDP transmissions are preferred as they do not require testing for
dropped packets and as such since each T.38 packet transmission includes
a majority of the data sent in the prior packet, a T.38 termination
point has a higher degree of success in re-assembling the fax
transmission back into its original form for interpretation by the end
device. This in an attempt to overcome the obstacles of simulating real
time transmissions using packet based protocol.
[30]
There have been updated versions of T.30 to resolve the fax over IP
issues, which is the core fax protocol. Some newer high end fax machines
have T.38 built-in capabilities which allow the user to plug right into
the network and transmit/receive faxes in native T.38 like the Ricoh
4410NF Fax Machine.
[31]
A unique feature of T.38 is that each packet contains a portion of the
main data sent in the previous packet. With T.38, two successive lost
packets are needed to actually lose any data. The data one will lose
will only be a small piece, but with the right settings and error
correction mode, there is an increased likelihood that they will receive
enough of the transmission to satisfy the requirements of the fax
machine for output of the sent document.
While many late-model
analog telephone adapters support T.38, uptake has been limited as many voice-over-IP providers perform
least-cost routing
which selects the least expensive PSTN gateway in the called city for
an outbound message. There is typically no means to ensure that that
gateway is T.38 capable. Providers often place their own equipment (such
as an
Asterisk PBX
installation) in the signal path, which creates additional issues as
every link in the chain must be T.38 aware for the protocol to work.
Similar issues arise if a provider is purchasing local
direct inward dial numbers from the lowest bidder in each city, as many of these may not be T.38 enabled.
Power requirements
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 electrical 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.
[32]
Some VoIP service providers use customer premises equipment (e.g.,
cablemodems) 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.
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.
[original research?]
However, the network design envisioned by DARPA in the early 1980s
included a fault tolerant architecture under adverse conditions.
Security
The security concerns of VoIP telephone systems are similar to those of any Internet-connected device. This means that
hackers who know about these vulnerabilities can institute
denial-of-service attacks, harvest customer data, record conversations and compromise voicemail messages.
[33][34][35]
Compromised VoIP user account or session credentials may enable an
attacker to incur substantial charges from third-party services, such as
long-distance or international telephone calling.
The technical details of many VoIP protocols create challenges in routing VoIP traffic through
firewalls and
network address translators, used to interconnect to transit networks or the Internet. Private
session border controllers
are often employed 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, enabling it to traverse
symmetric NATs and firewalls. Other methods to traverse NAT devices involve assistive protocols such as
STUN and
Interactive Connectivity Establishment (ICE).
Many consumer VoIP solutions do not support encryption of the
signaling path or the media, however securing a VoIP phone is
conceptually easier to implement than on traditional telephone circuits.
A result of the lack of encryption is a relative easy to eavesdrop on
VoIP calls when access to the data network is possible.
[36] Free open-source solutions, such as
Wireshark, facilitate capturing VoIP conversations.
Standards for securing VoIP are available in the
Secure Real-time Transport Protocol (SRTP) and the
ZRTP protocol for
analog telephony adapters as well as for some
softphones.
IPsec is available to secure
point-to-point VoIP at the transport level by using
opportunistic encryption.
In 2005, Skype invited a researcher, Tom Berson, to assess the security
of the Skype software, and his conclusions are available in a published
report.
[37]
Government and military organizations use various security measures
to protect VoIP traffic, such as voice over secure IP (VoSIP), secure
voice over IP (SVoIP), and secure voice over secure IP (SVoSIP).
[38] The distinction lies in whether encryption is applied in the telephone or in the network
[39] or both. 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.
[40][41][42][43][44] Public Secure VoIP is also available with free GNU programs and in many popular commercial VoIP programs via libraries such as
ZRTP.
[45]
Caller ID
Caller ID
support among VoIP providers varies, but is provided by the majority of
VoIP providers. Many VoIP service providers allow callers to configure
arbitrary caller ID information, thus permitting
spoofing attacks.
[46]
Business-grade VoIP equipment and software often makes it easy to
modify caller ID information, providing many businesses great
flexibility.
The United States enacted the
Truth in Caller ID Act of 2009
on December 22, 2010. This law makes it a crime to "knowingly transmit
misleading or inaccurate caller identification information with the
intent to defraud, cause harm, or wrongfully obtain anything of
value ...".
[47] Rules implementing the law were adopted by the
Federal Communications Commission on June 20, 2011.
[48]
Compatibility with traditional analog telephone sets
Most analog telephone adapters do not decode dial pulses generated by older telephones, supporting only
touch-tone. Pulse-to-tone converters are commercially available;
[49] user reports that a few specific ATA models (such as the
Grandstream 502) recognise pulse dial directly
[50][51] are poorly documented and provide no assurance that newer models in the same series will retain this compatibility.
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
consumers' houses.
User and administrative interfaces
Voice over IP services typically take advantage of other Internet- or
web-based facilities for operation and administration. Websites provide
customer interaction, account configuration, service statistics, and
billing. In addition, VoIP communication sessions may be launched
directly from web-pages or software that issue requests to web-based
facilities.
Web-based VoIP
uses this integration to conduct telephone sessions without the need
for a telephone set, be it conventional POTS- or IP-based. An example is
the
click-to-call
service, in which a software agent running in the web-browser permits
users to click on a telephone number embedded in any web page to
initiate a telephone call. The service only requires a microphone and an
audio head set connected to the user's computer.
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.[52]
- 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.
Regulatory and legal issues
As the popularity of VoIP grows, governments are becoming more
interested in regulating VoIP in a manner similar to PSTN services.
[53]
Throughout the developing world, countries where regulation is weak
or captured by the dominant operator, restrictions on the use of VoIP
are imposed, including in
Panama
where VoIP is taxed, Guyana where VoIP is prohibited and India where
its retail commercial sales is allowed but only for long distance
service.
[54] In
Ethiopia,
where the government is nationalising telecommunication service, it is a
criminal offence to offer services using VoIP. The country has
installed firewalls to prevent international calls being made using
VoIP. These measures were taken after the popularity of VoIP reduced the
income generated by the state owned telecommunication company.
European Union
|
|
This section is outdated. Please update this article to reflect recent events or newly available information.
Last update: 2006 (September 2013) |
In the
European Union,
the treatment of VoIP service providers is a decision for each national
telecommunications regulator, which must use competition law to define
relevant national 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).
[citation needed]
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.
[citation needed]
India
In
India, it is legal to use VoIP, but it is illegal to have
VoIP gateways inside India.
[55]
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 is not permitted by law to be inside India.
[55]
In the interest of the Access Service Providers and International
Long Distance Operators the Internet telephony was permitted to the ISP
with restrictions. Internet Telephony is considered to be different
service in its scope, nature and kind from real time voice as offered by
other Access Service Providers and Long Distance Carriers. Hence the
following type of Internet Telephony are permitted in India:
[56]
- (a) PC to PC; within or outside India
(b) PC / a device / Adapter conforming to standard of any international
agencies like- ITU or IETF etc. in India to PSTN/PLMN abroad.
(c) Any device / Adapter conforming to standards of International
agencies like ITU, IETF etc. connected to ISP node with static IP
address to similar device / Adapter; within or outside India.
(d) Except whatever is described in condition (ii) above, no other form of Internet Telephony is permitted.
(e) In India no Separate Numbering Scheme is provided to the Internet
Telephony. Presently the 10 digit Numbering allocation based on E.164 is
permitted to the Fixed Telephony, GSM, CDMA wireless service. For
Internet Telephony the numbering scheme shall only conform to IP
addressing Scheme of Internet Assigned Numbers Authority (IANA).
Translation of E.164 number / private number to IP address allotted to
any device and vice versa, by ISP to show compliance with IANA numbering
scheme is not permitted.
(f) The Internet Service Licensee is not permitted to have PSTN/PLMN
connectivity. Voice communication to and from a telephone connected to
PSTN/PLMN and following E.164 numbering is prohibited in India.
Middle East
In the
UAE and
Oman it is illegal to use any form of VoIP, to the extent that Web sites of
Gizmo5
are blocked. Providing or using VoIP services is illegal in Oman. Those
who violate the law stand to be fined 50,000 Omani Rial (about 130,317
US dollars) or spend two years in jail or both. In 2009, police in Oman
have raided 121 Internet cafes throughout the country and arrested 212
people for using/providing VoIP services.
[citation needed]
South Korea
In
South 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 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 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 on base may continue
to use their US-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 US VoIP providers.
[57]
United States
In the United States, the
Federal Communications Commission
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.
[58] 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.
[59]
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.
[60]
Pronunciation
The acronym
VoIP has been pronounced variably since the inception of the term. Apart from spelling out the acronym letter by letter,
vē'ō'ī'pē (
vee-oh-eye-pee), there are three likely possible pronunciations:
vō'ī'pē[needs IPA] (
vo-eye-pee) and
vō'ip[needs IPA] (
vo-ipp), have been used, but generally, the single syllable
vŏy'p[needs IPA] (
voyp, as in
voice) may be the most common within the industry.
[61]
Historical milestones
- 1973: Network Voice Protocol (NVP) developed by Danny Cohen and others to carry real time voice over Arpanet[citation needed]
- 1974: The Institute of Electrical and Electronic Engineers (IEEE) published a paper titled "A Protocol for Packet Network Interconnection".[62]
- 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 Cerf, 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.[63]
- 1986: Proposals from various standards organizations[specify] for Voice over ATM, in addition to commercial packet voice products from companies such as StrataCom
- 1991: First Voice Over IP application, Speak Freely, released as public domain. Originally written by John Walker and further developed by Brian C. Wiles.[64]
- 1992: Voice over Frame Relay standards development within Frame Relay Forum
- 1994: MTALK, a freeware VoIP application for Linux[65]
- 1995: VocalTec releases the first commercial Internet phone software.[66][67]
- 1996:
- ITU-T
begins development of standards for the transmission and signaling of
voice communications over Internet Protocol networks with the H.323 standard.[69]
- US telecommunication companies petition the US Congress to ban Internet phone technology.[70]
- 1997: Level 3 began development of its first softswitch, a term they coined in 1998.[71]
- 1999:
- 2004: Commercial VoIP service providers proliferate.
See also
References
of Italy and Mexico, in which case - carry on, folks. But if you think
differently - say, that Microsoft will benefit by having its services
available on lots of devices, but most of all by having them used on
lots of devices where it is the default - then let’s follow this.
