Voice over IP

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Image:Voip-typical.gif Image:Voip-box.gif Voice over Internet Protocol (also called VoIP, IP Telephony, Internet telephony, and Broadband Phone) is the routing of voice conversations over the Internet or any other IP-based network. The voice data flows over a general-purpose packet-switched network, instead of traditional dedicated, circuit-switched telephony transmission lines.

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 might be deployed on any IP network, including ones lacking a connection to the rest of the Internet, for instance on a private building-wide LAN.

Advantages

Cost

In general, phone service via VoIP 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 under-utilized network capacity they can use for VoIP at no additional cost. Some Internet connections are asymmetrical, i.e. the upstream data rate is significantly lower than the downstream data rate. This places a final absolute throttle to the transmitted data rate and thus voice quality. The slowest Internet connections can offer lower signal quality than regular dedicated phone networks.

VoIP to VoIP phone calls on any provider are typically free, whilst VoIP to PSTN calls generally costs the VoIP user. Free VoIP to PSTN services are rare. A notable provider is VoIP User.

There are two types of PSTN to VoIP services: DID 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 anywhere you connect it to the Internet, you can receive your incoming calls.
Call center agents using VoIP phones can work from anywhere with a sufficiently fast Internet connection.
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.

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 to make all of this possible.
Users of Instant Messenger based VoIP services like Skype, Gizmo Project or Yahoo! Messenger can also travel anywhere in the world and make and receive phone calls.

Drawbacks

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 legacy digital PBX ports.

Implementation challenges

Because IP does not provide any mechanism to ensure that data packets are delivered in sequential order, or provide any Quality of Service guarantees, VoIP implementations may face problems dealing with latency (especially if satellite circuits are involved), and jitter. They are faced with the problem of restructuring streams of received IP packets, which can come in any order and have packets delayed or missing, to ensure that the ensuing audio stream maintains a proper time consistency. This functionality is usually accomplished by means of a jitter buffer. Another main challenge is routing VoIP traffic to traverse certain firewalls and NAT. Intermediary devices called Session Border Controllers (SBC) are often used to achieve this, though some proprietary systems such as Skype traverse firewall and NAT without a SBC by using users' computers as super node servers to route other people's calls. Other methods to traverse firewalls involve using protocols such as STUN or ICE.

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, thus saving you money when you switch to VoIP. VoIP can also be used with Cable Internet instead of DSL, eliminating the need to purchase two telephone lines.

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. In order to use VoIP during a power outage, an uninterruptible power supply or a generator must be installed on the premises. 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.

Some broadband connections may have less than desirable reliability. 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. 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 commercial services such as Vonage, 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. Other services, such as Skype & PeerMe, typically 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, such as BroadVoice 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. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. There are several open source solutions like VoIPong or Vomit 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 by 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 usually not available at a consumer level.

In this context, 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. This information is correct as of April 2006.

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.

Adoption

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. This requires an analog telephone adapter (ATA) to connect a telephone to the broadband Internet connection. 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. One advantage of this 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.

For the present, the broadband phone is likely to complement, rather than replace, a PSTN line, due to a number of inconveniences compared to traditional services. It still needs a power supply, and ready access to a broadband Internet connection. 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.

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.

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 Voice over Internet Protocol, or the customer may request an immediate call back by entering their phone number. One signficiant 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.

Legal

Template:Expand 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 PSTN services.

In the U.S., the Federal Communications Commission now requires all VoIP operators who don't 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 [1]. The Telecommunications Act of 2005 proposes adding more traditional PSTN regulations, such as local number portability and universal service fees. Other future legal issues include wiretapping and network neutrality.

Some Latin American countries, fearful for their state owned telephone services, have placed restrictions on the use of VoIP, including in Panama where VoIP is taxed. In Ethiopia, where the telecommunication service is monopolized by a single service provider, it is a criminal offence to give services in 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 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): - the former 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; - the latter 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". As concerns obligations which can exist independently of market power (e.g. the obligation to offer access to emergency calls), the relevant EU Directive is unfortunately terribly drafted and it is impossible to say definitely 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.

Technical details

There is a lot of debate over the two most popular types of VoIP: SIP and 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 500-600ms) 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 5.6kbps compressed audio path will require 18kbps of bandwidth based on standard sampling rates. The difference between the 5.6kbps and 18kbps 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 6-7kbps 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.

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 Mitel
CorNet-IP: proprietary protocol from Siemens
IAX: the Inter-Asterisk eXchange protocol used by the Asterisk open source PBX server 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.

See also

Networks: Full Phone Service

AOL TotalTalk : A U.S. and Canadian VoIP phone service provider.
AT&T CallVantage : A U.S.-based Business VoIP carrier.
Bandwidth.com : A U.S.-based Business VoIP carrier.
Blueface : An Ireland-based VoIP provider
BroadVoice : A U.S.-based VoIP network.
BT Broadband Voice : A UK based VoIP service provided through BT
Cliconnect Internet Telephony : A Canadian-Brazilian based Voip Service Provider.
Congruent IP Communications : A Canadian-based hosted VoIP services provider.
Covad : A U.S.-based hosted VoIP services provider.
Globe7 : An emerging service teamed with Lycos
Internode NodePhone : An Australian-based VoIP service provider.
Iristel Inc. : A Canadian based VoIP service provider with Asterisk and T.38 fax support.
IPvaani : IPvaani is free VoIP network service provider for its subscribers.
IXP Voice : A U.S.-based VoIP phone service provider.
Lingo : U.S. VoIP offering of Primus Telecommunications.
OpenWengo : A France-based VoIP Provider.
DonauTel : An Austrian (EU)-based VoIP Provider.
MetroTel : A U.S.-based VoIP Provider.
MyWebCalls : A UK-based VoIP phone service.
Net2Phone : A U.S.-based VoIP pioneer founded in 1996.
Packet8 : A U.S.-based VoIP network.
sipgate: German SIP based provider also operating in Austria and the UK
SIPphone : A SIP-based VoIP network.
SunRocket : A U.S.-based VoIP phone service provider.
TheGlobe : A U.S.-based Voice over Internet Protocol communications company.
TeleCable Services : A U.S.-based VoIP phone service provider.
Telio : A Norwegian-based VoIP phone service provider
TexVoIP : Italian VoIP phone service provider
Thinking Phone Networks: A US based business VoIP systems & service provider
VoicePulse : A U.S.-based VoIP phone service provider
Vonage : A U.S.-based VoIP phone service provider
Verizon VoiceWing : Verizon's version of VoIP
VoiceEclipse : US LEC's U.S.-based VoIP retail & wholesale phone service.
Voicestick : A global VoIP network service that also provides cellular bridge, VoIP chat, VoIP video, email based VoIP voicemail and VoIP e911 services
Voip Stunt : A provider which lets you make free VOIP-Phone calls to some countries.
VoIP User : A UK based free SIP based network with a community funded PSTN gateway
Voxbone : A EU based provider of worldwide VoIP virtual numbers

Networks: Software Service

Camfrog : Camfrog Video Chat uses a proprietary system with instant messaging to do multi-user audio and video.
FWD (formerly Free World Dialup) : A free SIP-based VoIP network.
Gizmo_Project : Gizmo Project uses your internet connection (broadband or dial-up) to make calls to other computers, phones and mobiles.
Paltalk : A proprietary freeware VoIP system which uses a messenger-like client for video, voice and text chat. Also offers interoperable Instant messaging (IM) with AOL Instant Messenger , Yahoo Messenger, MSN Messenger and ICQ using the Paltalk client.
PeerMe : A proprietary freeware VoIP system which uses a messenger-like client.
Skype : A proprietary freeware VoIP system which uses a messenger-like client.
SIP Broker : One of the biggest free VoIP peering and ENUM services.
Teleo : A VoIP network using a P2P model
TelSIP : A European-based VoIP using SIP.
TheGlobe : A proprietary freeware VoIP plugin which adds a messenger-like client to your browser.
Vbuzzer : A SIP-based VoIP service with low cost access to conventional PSTN network.
Chattercube : The worlds first Peer-to-Peer VoIP network.

Hardware: Wi-Fi Phone

UTStarcom F1000: a wi-fi SIP handset
Cisco IP Phone 7960 series: a VoIP handset for use in offices

Software

Asterisk PBX : The popular Linux-based open source software PBX switch. It is known to compile and run on GNU/Linux for PPC along with OpenBSD, FreeBSD, and Mac OS X Jaguar.
Cellfire Security : a vendor of mobile VoIP security software. The secure agent encrypts end-to-end a phone call.
GNU Bayonne : A free telecommunications application server written for the GNU project.
Ekiga : The popular Linux-based open source softphone which supports a wide variety of video and audio codecs. Version 1 supported only H.323, while version 2 supports H.323 and SIP.
GameComm, Roger Wilco, Teamspeak, and Ventrilo : Voice communication programs popular in online gaming.
Gizmo : A freeware VoIP client using SIP with Jabber protocol support.
Google Talk : A free VoIP system from Google.
IP Multimedia Subsystem : architectural model (with several SIP extensions), used by the traditional telecommunications industry to develop systems to replace the current circuit switched network with a NGN network.
Jajah : A freeware VoIP client with free videotelephony, chat, text messaging, voicemailbox and is compatible to SIP, Skype, Gizmo and IAX/H.323
KCall is an open source Linux softphone using Qt toolkit libraries.
MSN Messenger
OpenH323 : An implementation of the H.323 protocol released under the MPL license. It is used as the basis of several other VoIP applications including Ekiga and GnuGK. A new version of OpenH323 called OPAL is available that supports both H.323 and SIP
OpenWengo : Voip-application published under GNU GPL license
PeerMe : A proprietary freeware VoIP system which uses a messenger-like client.
PGPfone: An older secure voice system based on the popular PGP encryption package. No longer being developed. See Zfone instead.
PhoneGaim : A free VoIP system based on Gaim and SIP.
PortaOne : Billing Solutions for VOIP www.portaone.com.
ReSIProcate : A robust and feature-rich open source SIP stack.
Scopserv : Web Management GUI for Asterisk PBX
SIMPLE : An instant messaging and presence protocol based on SIP.
SIP Express Router (SER) : Fast, scalable, free SIP server.
sipX : The popular open source SIP PBX, native SIP call control, many features, Web management, and fully standards-compliant
SJphone : SJphone is a popular free SIP/H.323 softphone that many services use.
Skype : Skype is a free VoIP client that offers in and outbound PSTN facilities. It is closed source and is based on a closed protocol.
*starShop : Open Source professional and powerful billing and management system based on Asterisk PBX for Calling Shops and Internet cafes.
TERAVoice Server - TERAVoice VoIP Gateway
Tivi : A SIP VoIP client softphone.
Vbuzzer : Vbuzzer Softphone is a proprietary freeware to be used in conjunction with Vbuzzer Internet telephone service.
Yahoo! Messenger
YATE : A GPL (free) software VoIP telephony engine (VoIP server and client for H.323,IAX,SIP) for Windows and Linux
Zfone: a secure VoIP open source VoIP system by Phil Zimmermann, the creator of the PGP encryption software.

VoIP Testing

SIP_Forum: The SIP Forum organizes sipit events two times a year, used by vendors to assure interoperability of products.
TestYourVoIP: A free VoIP quality test website (requires a Java-enabled Web browser).

VOIP Test Equipment Vendors

Related concepts

External links and research sources

ca:VoIP cs:VoIP da:VoIP de:IP-Telefonie es:Voz sobre IP fi:VoIP fr:Voice over IP he:טלפון מבוסס אינטרנט id:Voice over IP it:Voice over IP hu:Voice over IP ja:IP電話 lt:IP telefonija nl:Voice over IP pl:VoIP pt:Voz sobre IP ru:VoIP sv:IP-telefoni sl:VoIP th:วีโอไอพี zh:IP电话 zh:網路電話

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