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.
Fixed delays cannot be controlled
but some delays can be minimized by marking voice packets as being
delay-sensitive (see, for example, Diffserv).
The principal cause of packet
loss is congestion, which can be controlled by congestion management
and avoidance. Carrier voip networks avoid congestion by means
of teletraffic engineering.
Variation in delay is called jitter.
The effects of jitter can be mitigated by storing voice packets
in a buffer (called a play-out buffer) upon arrival, before playing
them out. This avoids a condition known as buffer underrun, in
which the playout process runs out of voice data to play because
the next voice packet has not yet arrived, but increases delay
by the length of the buffer.
Common causes of echo include
impedance mismatches in analog circuitry, and acoustic coupling
of the transmit and receive signal at the receiving end.
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 is 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, the 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
A number of protocols have been
defined to support the reporting of QoS/QoE for voip calls. These
include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex
B (for H.323), H.248.30 and MGCP extensions. The RFC3611 voip
Metrics block is generated by an IP phone or gateway during a
live call and contains information on packet loss rate, packet
discard rate (due to jitter), packet loss/discard burst metrics
(burst length/density, gap length/density), network delay, end
system delay, signal / noise / echo level, MOS scores and R factors
and configuration information related to the jitter buffer.
RFC3611 voip metrics reports are
exchanged between IP endpoints on an occasional basis during a
call, and an end of call message sent via SIP RTCP Summary Report
or one of the other signaling protocol extensions. RFC3611 voip
metrics reports are intended to support real time feedback related
to QoS problems, the exchange of information between the endpoints
for improved call quality calculation and a variety of other applications.
Difficulty with sending faxes
The support of sending faxes over
voip is still limited. The existing voice codecs are not designed
for fax transmission. An effort is underway to remedy this by
defining an alternate IP-based solution for delivering Fax-over-IP,
namely the T.38 protocol. Another possible solution to overcome
the drawback is to treat the fax system as a message switching
system which does not need real time data transmission - such
as sending a fax as an email attachment (see Fax) or remote printout
(see Internet Printing Protocol). The end system can completely
buffer the incoming fax data before displaying or printing the
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.
Moreover, in the event that the
caller is unable to give an address, emergency services may be
unable to locate them in any other way. Following the lead of
mobile phone operators, several voip carriers are already implementing
a technical work-around.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.
Conventional phones are connected
directly to telephone company phone lines, which in the event
of a power failure are kept functioning by back-up generators
or batteries located at the telephone exchange. However, household voip hardware uses broadband modems and other equipment powered
by household electricity, which may be subject to outages dictating
the use of an uninterruptible power supply or generator to ensure
availability during power outages. 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 extremely reliable,
most broadband networks are less than 10 years old, and even the
best are still subject to intermittent outages. See call control.
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.
Integration into global telephone
While the traditional Plain Old
Telephone System (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.
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.
In a few cases, voip providers
may allow a caller to spoof the Caller ID information, making
it appear as though they are calling from a different number.
Business grade voip equipment and software often makes it easy
to modify caller ID information. Although this can provide many
businesses great flexibility, it is also open to abuse.
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.However, note that many hotspots
require browser-based authentication, which most SIP phones do
The 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. 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 ex-pats,
who either, due to not having a fixed or mobile phone or high
overseas roaming charges, choose instead to use voip services
to make their phone calls. Pre-paid phone cards can be used either
from a normal phone or from Internet cafes that have phone services.
Developing countries and areas with high tourist or immigrant
communities generally have a higher uptake.
Voice over Instant Messenger,
like popular Skype, Voice over MSN, Yahoo, QQ in China and Google
Talk. VoIM is one kind of general voip that was based on an IM.
voip, specifically, usually is referred as traditional SIP or
H.323 IP phone, as opposed to VoIM as newly emerged Skype-like
A major development starting in
2004 has been the introduction of mass-market voip services over
broadband Internet access services, in which subscribers make
and receive calls as they would over the PSTN. Full phone service
voip phone companies provide inbound and outbound calling with
Direct Inbound Dialing. Many offer unlimited calling to the U.S.,
and some to Canada or selected countries in Europe or Asia as
well, for a flat monthly fee.
These services take a wide variety
of forms which can be more or less similar to traditional POTS.
At one extreme, an analog telephone adapter (ATA) may be connected
to the broadband Internet connection and an existing telephone
jack in order to provide service nearly indistinguishable from
POTS on all the other jacks in the residence. This type of service,
which is fixed to one location, is generally offered by broadband
Internet providers such as cable companies and telephone companies
as a cheaper flat-rate traditional phone service. Often the phrase
"voip" is not used in selling these services, but instead the
industry has marketed the phrases "Internet Phone", "Digital Phone"
or "Softphone" which is aimed at typical phone users who are not
necessarily tech-savvy. Typically, the provider touts the advantage
of being able to keep one's existing phone number.
At the other extreme are services
like Gizmo Project and Skype which rely on a software client on
the computer in order to place a call over the network, where
one user ID can be used on many different computers or in different
locations on a laptop. In the middle lie services which also provide
a telephone adapter for connecting to the broadband connection
similar to the services offered by broadband providers (and in
some cases also allow direct connections of SIP phones) but which
are aimed at a more tech-savvy user and allow portability from
location to location. One advantage of these two types of services
is the ability to make and receive calls as one would at home,
anywhere in the world, at no extra cost. No additional charges
are incurred, as call diversion via the PSTN would, and the called
party does not have to pay for the call. For example, if a subscriber
with a home phone number in the U.S. or Canada calls someone else
within his local calling area, it will be treated as a local call
regardless of where that person is in the world. Often the user
may elect to use someone else's 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 problems, but in other cases they will not go through
at all. And in some cases, this equipment can be made to work
over a voip connection if the sending speed can be changed to
a lower bits per second rate. If voip and cellular substitution
becomes very popular, some ancillary equipment makers may be forced
to redesign equipment, because it would no longer be possible
to assume a conventional voice-grade telephone line would be available
in almost all homes in North America and Western-Europe. The TestYourvoip
website offers a free service to test the quality of or diagnose
an Internet connection by placing simulated voip calls from any
Java-enabled Web browser, or from any phone or voip device capable
of calling the PSTN network.
Mobile phones & Handheld
Telcos and consumers have invested
billions of dollars in mobile phone equipment. In developed countries,
mobile phones have achieved nearly complete market penetration,
and many people are giving up landlines and using mobiles exclusively.
Given this situation, it is not entirely clear whether there would
be a significant higher demand for voip among consumers until
either public or community wireless networks have similar geographical
coverage to cellular networks (thereby enabling mobile voip phones,
so called WiFi phones or VoWLAN) or voip is implemented over legacy
3G networks. However, "dual mode" telephone sets, which allow
for the seamless handover between a cellular network and a WiFi
network, are expected to help voip become more popular.
Phones like the NEC N900iL, and
later the Nokia E60, E61 have been the first "dual mode" telephone
sets capable of delivering mobile voip. With more and more mobile
phones and handheld devices using voip, the nicknames of "MoIP"
and Mvoip (Mobile voip)have been attributed to these mobile applications.
Handheld Devices are another type
of medium whereby you can use voip services. Since most of these
devices are limited to using GSM/GPRS type of communication mediums,
almost all of the handheld devices use WiFi of some sort.
Another addition to handheld devices
are ruggedized barcode 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
Corporate and telco use
Although few office environments
and even fewer homes use a pure voip infrastructure, telecommunications
providers routinely use IP telephony, often over a dedicated IP
network, to connect switching stations, converting voice signals
to IP packets and back. The result is a data-abstracted digital
network which the provider can easily upgrade and use for multiple
Corporate customer telephone support
often use IP telephony exclusively to take advantage of the data
abstraction. The benefit of using this technology is the need
for only one class of circuit connection and better bandwidth
use. Companies can acquire their own gateways to eliminate third-party
costs, which is worthwhile in some situations.
voip is widely employed by carriers,
especially for international telephone calls. It is commonly used
to route traffic starting and ending at conventional PSTN telephones.
Many telecommunications companies
are looking at the IP Multimedia Subsystem (IMS) which will merge
Internet technologies with the mobile world, using a pure voip
infrastructure. It will enable them to upgrade their existing
systems while embracing Internet technologies such as the Web,
email, instant messaging, presence, and video conferencing. It
will also allow existing voip systems to interface with the conventional
PSTN and mobile phones.
Click to call
Click-to-call is a service which
lets users click a button and immediately speak with a customer
service representative. The call can either be carried over voip,
or the customer may request an immediate call back by entering
their phone number. One significant benefit to click-to-call providers
is that it allows companies to monitor when online visitors change
from the website to a phone sales channel.
As the popularity of voip grows,
and PSTN users switch to voip in increasing numbers, governments
are becoming more interested in regulating VoIP VoIP in a manner
similar to legacy PSTN services, especially with the encouragement
of the state-mandated telephone monopolies/oligopolies in a given
country, who see this as a way to stifle the new competition.
In the European Union, the treatment
of voip service providers is a decision for each Member State's
national telecoms 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).
voip services that function over
managed networks are often considered to be a viable substitute
for PSTN telephone services (despite the problems of power outages
and lack of geographical information); as a result, major operators
that provide these services (in practice, incumbent operators)
may find themselves bound by obligations of price control or accounting
voip services that function over
unmanaged networks are often considered to be too poor in quality
to be a viable substitute for PSTN services; as a result, they
may be provided without any specific obligations, even if a service
provider has "significant market power".
The relevant EU Directive is not
clearly drafted concerning obligations which can exist independently
of market power (e.g., the obligation to offer access to emergency
calls), and it is impossible to say definitively whether voip
service providers of either type are bound by them. A review of
the EU Directive is under way and should be complete by 2007.