Voice over IP
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A typical analog telephone adapter for connecting an ordinary phone to a VoIP network |
Voice over Internet Protocol, also called
VoIP (pronounced "vee-oh-eye-pee" [
1] or "voyp"),
IP Telephony,
Internet telephony,
Broadband telephony,
Broadband Phone and
Voice over Broadband is the
routing of
voice conversations over the
Internet or through any other
IP-based network.
Protocols used to carry voice signals over the IP network are commonly referred to as
Voice over IP or
VoIP protocols. They may be viewed as commercial realizations of the experimental
Network Voice Protocol (
1973) invented for the
ARPANET.
Voice over IP traffic can be deployed on any IP network, including those lacking a connection to the rest of the Internet, for instance on a
local area network.
Cost
In general, phone service via VoIP is free or costs less than equivalent service from traditional sources but similar to alternative Public Switched Telephone Network (
PSTN) service providers. Some cost savings are due to using a single network to carry voice and data, especially where users have existing underutilized network capacity they can use for VoIP at no additional cost. VoIP to VoIP phone calls on any provider are typically free, whilst VoIP to PSTN calls generally costs the VoIP user.
There are two types of PSTN to VoIP services:
DID (Direct Inward Dialing) and
access numbers. DID will connect the caller directly to the VoIP user while access numbers requires the caller to input the extension number of the VoIP user. Access numbers are usually charged as a local call to the caller and free to the VoIP user while DID usually has a monthly fee. There are also DID that are free to the VoIP user but is chargeable to the caller.
Functionality
VoIP can facilitate tasks that may be more difficult to achieve using traditional phone networks:
*Incoming phone calls can be automatically routed to your VoIP phone, regardless of where you are connected to the network. Take your VoIP phone with you on a trip, and wherever you connect to the Internet, you can receive incoming calls.
*Free phone numbers for use with VoIP are available in the USA, UK and other countries from organizations such as
VoIP User.
*Call center agents using VoIP phones can work from anywhere with a sufficiently fast Internet connection.
* Many VoIP packages include PSTN features that most telcos normally charge extra for, or may be unavailable from your local telco, such as 3-way calling, call forwarding, automatic redial, and caller ID.
Mobility
VoIP allows users to travel anywhere in the world and still make and receive phone calls:
*Subscribers of phone-line replacement services can make and receive local phone calls regardless of their location. For example, if a user has a New York City phone number and is traveling in Europe and someone calls the phone number, it will ring in Europe. Conversely, if a call is made from Europe to
New York City, it will be treated as a local call. Of course, there must be a connection to the Internet e.g.
WiFi to make all of this possible.
*Users of
Instant Messenger based VoIP services can also travel anywhere in the world and make and receive phone calls.
*VoIP phones can integrate 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.
VoIP technology still has a few shortcomings that have led some to believe that it is not ready for widespread deployment. However, many industry analysts predicted that
2005 was the "Year of Inflection," where more IP PBX ports shipped than conventional digital
PBX ports.
Drawbacks:One drawback is the inability to send faxes due to software and networking restraints in most home systems. However, an effort is underway to define an alternate IP-based solution for delivering Fax-over-IP, namely the
T.38 protocol.Another drawback is the inability to make phone calls during a power outage but this problem also exists with many phonesused with conventional land lines and can be remedied with a battery backup.If VoIP is used in solitary LAN (with no internet connection), it would consume more resources compared to a PABX.
Because IP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide
Quality of Service guarantees, VoIP implementations face problems dealing with
latency and
jitter. This is especially true when satellite circuits are involved. 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.
Another challenge is routing VoIP traffic through
firewalls and
address translators. Private
Session Border Controllers are used along with firewalls to enable VoIP calls to and from a protected enterprise network.
Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse
symmetric NATs and firewalls. Other methods to traverse firewalls involve using protocols such as
STUN or
ICE.
VOIP challenges:
*Delay/Network Latency
*Packet loss.
*Jitter.
*Echo.
*Security.
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
traffic 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.
DSL Internet access
VoIP technology does not necessarily require broadband Internet access, but this usually supports better quality of service. A sizable percentage of homes today are connected to the Internet through
DSL, which requires a traditional phone line. Having to pay for VoIP in addition to both a basic phone line and broadband Internet access reduces the potential benefits of VoIP. However, some regional telephone companies now offer DSL service without the phone (often called "
naked DSL" or "dry loop DSL"), thus saving subscribers money when they switch to VoIP. VoIP can also be used with
Cable Internet instead of DSL, potentially eliminating the need for a traditional phone line entirely.
Reliability
Conventional telephones 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. Furthermore, consumer network technologies such as cable and DSL often are not subject to the same restoration service levels as the PSTN or business technologies such as T-1 connection.
Quality of Service
Some broadband connections may have less than desirable quality. Where IP packets are lost or delayed at any point in the network between VoIP users, there will be a momentary drop-out of voice. This is more noticeable in highly congested networks and/or where there 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.
Emergency calls
The nature of
IP makes it difficult to geographically locate network users.
Emergency calls, therefore, cannot easily be routed to a nearby call center, and are impossible on some VoIP systems. 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 the VOIP line will be used at. 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.
The
United States government had set a deadline, requiring VoIP carriers to implement
E911; however, the deadline is being appealed by several of the leading VoIP companies.
This is a different situation with
IPBX systems, where these corporate systems often have full
E911 capabilities built into the system.
Integration into global telephone number system
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.
Single point of calling
With hardware VoIP solution 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 handsets 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.
Mobile phones
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 a) public or community
wireless networks have similar geographical coverage to cellular networks (thereby enabling mobile VoIP phones, so called WiFi phones) or b) VoIP is implemented over legacy
3G networks. However, "dual mode" handsets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular.
Security
The majority of consumer VoIP solutions do not support encryption yet. 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 is available on consumer devices from some manufacturers like
Sipura/Linksys for
Analog Telephone Adapters(ATAs) and
Gizmo Project for softphones (PCs/laptops emulating a phone).
The
Voice VPN solution provides
secure voice for enterprise VoIP networks by applying
IPSec encryption to the digitized voice stream.
As of April 2006, the
beta testing of
Zfone, a 'security wrapper' for certain VoIP systems by the inventor of
PGP, is notable, as a means by which strong security may be added to certain otherwise less secure VoIP systems. The softphone
Skype claims to use strong encryption by default, although it is not clear which encryption standards it uses.
Pre-Paid Phone Cards
VoIP has become a major provider of phone services to travellers, 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. The undeveloped markets are usually markets where Pre-Paid cards are used; however in cities with high tourist or immigrant communities they are also common.
Caller ID
Caller ID support among VoIP providers varies. When calling a PSTN number from some VoIP providers, Caller ID isn't supported, and the target person will not know who is calling. The number shows up as 'Unknown' or '000-012-3456'.
In a few cases, VoIP providers may allow a caller to
spoof the Caller ID information, making it appear that they are calling from a different number.
But the majority of VOIP providers now offer full Caller ID w/ Name on Outgoing calls.
Mass-market telephony
A major development starting in 2004 has been the introduction of
mass-market VoIP services over
broadband Internet access services, in which subscribers make and receive calls as they would over the
PSTN. Full phone service VoIP phone companies provide inbound and outbound calling with
Direct Inbound Dialing. Many offer unlimited calling to the U.S., and some to
Canada or selected countries in
Europe or
Asia as well, for a flat monthly fee.
These services take a wide variety of forms which can be more or less similar to traditional
POTS. At one extreme, an analog telephone adapter (ATA) may be connected to the broadband Internet connection and an existing telephone jack in order to provide service nearly indistinguishable from POTS on all the other jacks in the residence. This type of service, which is fixed to one location, is generally offered by broadband Internet providers such as cable companies and telephone companies as a cheaper flat-rate traditional phone service. Often the phrase "VoIP" is not used in selling these services, but instead the industry has marketed the phrase "Internet Phone" or "Digital Phone" 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. Examples of this type of service in the United States include
Time Warner and
Comcast's Digital Phone,
Verizon VoiceWing, and
AT&T CallVantage.
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 like
Vonage or
BroadVoice 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 a U.S.
area code calls someone else in his home area code, it will be treated as a local call regardless of where that person is in the world. Often the user may also select a phone number with any desired
area code; this is generally done to minimize the phone tariffs of those who frequently call.
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
911 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 particularly true for users who select a number with an area code outside their area.
Another challenge for these services is the proper handling of outgoing calls from
fax machines,
TiVo/
ReplayTV boxes,
satellite television receivers,
alarm systems, conventional
modems or FAXmodems, and other similar devices that depend on access to a voice-grade
telephone line for some or all of their functionality. At present, these types of calls sometimes go through without any problems, but in other cases they will not go through at all. And in some cases, this equipment can be made to work over a VoIP connection if the sending speed can be changed to a lower
bits per second rate. If VoIP and
cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional voice-grade telephone line would be available in almost all homes in North America and Western-Europe. The
TestYourVoIP website offers a free service to test the quality of or diagnose an Internet connection by placing simulated VoIP calls from any
Java-enabled Web browser, or from any phone or VoIP device capable of calling the PSTN network.
Corporate and telco use
Although few office environments and even fewer homes use a pure VoIP infrastructure, telecommunications providers routinely use IP telephony, often over a dedicated IP network, to connect switching stations, converting voice signals to IP packets and back. The result is a data-abstracted digital network which the provider can easily upgrade and use for multiple purposes.
Corporate customer telephone support often use IP telephony exclusively to take advantage of the data abstraction.The benefit of using this technology is the need for only one class of circuit connection and better bandwidth use. Companies can acquire their own gateways to eliminate third-party costs, which is worthwhile in some situations.
VoIP is widely employed by carriers, especially for international telephone calls. It is commonly used to route traffic starting and ending at conventional PSTN telephones.
Many telecommunications companies are looking at the
IP Multimedia Subsystem (IMS) which will merge Internet technologies with the mobile world, using a pure VoIP infrastructure. It will enable them to upgrade their existing systems while embracing Internet technologies such as the Web, email, instant messaging, presence, and video conferencing. It will also allow existing VoIP systems to interface with the conventional PSTN and
mobile phones.
Electronic Numbering (Enum) uses standard phone numbers (
E.164), but allows connections entirely over the Internet. If the other party uses Enum, the only expense is the Internet connection.
Use in Amateur Radio
Amateur radio has adopted VoIP by linking repeaters and users with
Echolink,
IRLP,
Dstar and
EQSO.
Echolink and
IRLP are programs/systems based upon the
Speak Freely VoIP open source software. In fact,
Echolink allows users to connect to repeaters via their computer (over the internet) rather than by using a radio. By using VoIP Amateur Radio operators are able to create large repeater networks with repeaters all over the world where operators can access the system with actual ham radios.
Ham Radio operators using radios are able to tune to repeaters with VoIP capabilities and use
DTMF buttons to command the repeater to connect to various other repeaters, thus allowing them to talk to people all around the world, however powerful their radio. Dingotel offers a similar feature for non ham radio users by providing a P2P network to link
FRS radios.
Click to call
Click-to-call is a service which lets users click a button and immediately speak with a customer service representative. The call can either be carried over VoIP, or the customer may request an immediate call back by entering their phone number. One significant benefit to click-to-call providers is that it allows companies to monitor when online visitors change from the website to a phone sales channel.
As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers, governments are becoming more interested in regulating VoIP in a manner similar to legacy PSTN services.
In the U.S., the
Federal Communications Commission now requires all VoIP operators who do not support
Enhanced 911 to attach a sticker warning that traditional 911 services aren't available. The FCC recently required VoIP operators to support
CALEA wiretap functionality. The
Telecommunications Act of 2005 proposes adding more traditional PSTN regulations, such as
local number portability and
universal service fees. Other future legal issues are likely to include laws against
wiretapping and
network neutrality.
Some
Latin American countries, fearful for their state owned telephone services, have imposed restrictions on the use of VoIP, including in
Panama where VoIP is taxed. In
Ethiopia, where a totalitarian government is monopolizing 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 a popularity in VoIP reduced the income generated by the state owned telecommunication company.
In the European Union, the treatment of VoIP service providers is a decision for each Member State's national telecoms regulator, which must use competition law theory 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 separation.
VoIP services that function over unmanaged networks are often considered to be too poor in quality to be a viable substitute for PSTN services; as a result, they may be provided without any specific obligations, even if a service provider has "significant market power".
The relevant EU Directive is not clearly drafted concerning obligations which can exist independently of market power (e.g. the obligation to offer access to emergency calls), and it is impossible to say definitively whether VoIP service providers of either type are bound by them. A review of the EU Directive is under way and should be complete by 2007.
In
India, it is legal to use VoIP. But it is illegal to have VoIP gateways inside
India. This effectively means, people who have PCs can use it to make a VoIP call to any number. But if the remote side is a normal phone, the gateway that converts VoIP call to
POTS call should not be inside
India.
The two major competing standards for VoIP are the IETF standard SIP and the ITU standard H.323. Initially H.323 was the most popular protocol, though its popularity has decreased in the "local loop" due to its poor traversal of NAT and firewalls. For this reason as domestic VoIP services have been developed, SIP has been far more widely adopted. However in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones, and the vast majority of callers have little or no idea that their POTS calls are being terminated over VoIP. So really SIP is a useful tool for the "local loop" and H.323 is like the "fiber backbone". With the most recent changes introduced for H.323, however, it is now possible for H.323 devices to easily and consistently traverse NAT and firewall devices, opening up the possibility that H.323 may again be looked upon more favorably in cases where such devices encumbered its use previously.
Where VoIP travels through multiple providers' Soft Switches the concept of Full Media Proxy and signalling proxy are important. In H.323 the data is made up of 3 streams of data: 1)
H.225.0 Call Signalling 2)
H.245 3) Media. So if you are in London, your provider is in Australia, and you wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600 ms) and packet loss will be high. However in signalling proxy mode where only the signalling flows through the provider the delay will be reduced to a more user friendly 120-150 ms. These proxy concepts could lead the way to true global providers.
One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically to send a G.723.1 5.6 kbit/s compressed audio path will require 18 kbit/s of bandwidth based on standard sampling rates. The difference between the 5.6 kbit/s and 18 kbit/s is packet headers. There are a number of bandwidth optimisation techniques used such as silence suppression and header compression. This can typically save 35% on bandwidth used. But the really interesting technology comes from VoIP off shoots such as TDMoIP which take advantage of the concept of bundling conversations that are heading to the same destination and wrapping them up inside the same packets. These can offer near toll quality audio in a 6-7 kbit/s data stream.
Protocols
Most standards-based solutions use either the
H.323 or
Session Initiation Protocol (SIP) protocols. A number of proprietary designs also exist. These typically support features such as call waiting, conference calling, and call transfer.
Transport protocols:;
SRTP : Existing secure transport protocol
RTP : Unsecure transport protocol;
ZRTP : New secure transport protocol proposal
Signaling protocols:;
Session Initiation Protocol (SIP) : defined by the
IETF, newer than H.323
H.323 : defined by the ITU-T;
Megaco (a.k.a. H.248) and
MGCP : both
media gateway control protocols
Skinny Client Control Protocol : proprietary protocol from Cisco;
MiNET : proprietary protocol from
MitelCorNet-IP : proprietary protocol from Siemens;
IAX : the Inter-Asterisk eXchange protocol used by the
Asterisk open source
PBX server,
Yate and associated client software
Skype : a proprietary peer-to-peer protocol used in the Skype application;
Jajah : a proprietary
peer-to-peer protocol used in the
Jajah SIP and IAX compatible webphone
Jingle : open peer-to-peer protocol based on XMPP (Jabber) and being harmonised with the 'substantially equivalent' Google Talk protocol.
Several different speech codecs can be used for stream audio compression. Commonly used codecs for VoIP traffic include G.711, G.723.1 and G.729, all ITU-T-specified.*
Computer conferencing*
Comparison of VoIP software*
Differentiated services*
Integrated services*
Predictive dialers*
Secure telephone*
SIP*
Mobile VoIP
*
FCC VoIP Information from the
FCC*
DMOZ listing of VoIP web sites*
VOIP Protocols and Standards
