NationMaster - Encyclopedia: Voice over Internet Protocol

A Voice over Internet Protocol (VoIP) is a protocol optimized for transmission of voice through the Internet or other packet switched networks. VoIP is often used abstractly to refer to the actual transmission of voice (rather than the protocol implimenting it). VoIP is also known as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband. "VoIP" is sometimes pronounced voyp. Image File history File links No higher resolution available. ... Image File history File links No higher resolution available. ... Image File history File links VoIP_analog_telephone_adapter_diagram. ... Image File history File links VoIP_analog_telephone_adapter_diagram. ... For other senses of this word, see protocol. ... In computer networking and telecommunications, packet switching is a communications paradigm in which packets (messages or fragments of messages) are individually routed between nodes, with no previously established communication path. ... A computer network is an interconnection of a group of computers. ... Broadband in telecommunications is a term that refers to a signaling method that includes or handles a relatively wide range of frequencies, which may be divided into channels or frequency bins. ...

Companies providing VoIP service are commonly referred to as providers, and protocols which are 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 providers. Some cost savings are due to utilizing a single network to carry voice and data, especially where users have existing underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to public switched telephone networks, PSTN, may have a cost that is borne by the VoIP user. This article concerns communication between pairs of electronic devices. ... The Network Voice Protocol (NVP) was a pioneering computer network protocol for transporting human speech over packetized communications networks. ... For the song by James Blunt, see 1973 (song). ... ARPANET logical map, March 1977. ... The public switched telephone network (PSTN) is the concatenation of the worlds public circuit-switched telephone networks, in much the same way that the Internet is the concatenation of the worlds public IP-based packet-switched networks. ...

Voice over IP protocols carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques, encapsulated in a data packet stream over IP. Digital audio comprises audio signals stored in a digital format. ... Speech coding is the application of data compression of digital audio signals containing speech. ... In most birds and reptiles, an egg (Latin ovum) is the zygote, resulting from fertilization of the ovum. ...

There are two types of PSTN to VoIP services: Direct Inward Dialing (DID) and access numbers. DID will connect the caller directly to the VoIP user while access numbers require the caller to input the extension number of the VoIP user. Direct Inward Dialing (DID, also called DDI in Europe) is a feature offered by telephone companies for use with their customers PBX systems, whereby the telephone company (telco) allocates a range of numbers all connected to their customers PBX. As calls are presented to the PBX, the number that...

History

Voice over IP has been a subject of interest almost since the first computer network. By 1973, voice was being transmitted over the early Internet.^[1] By Technology for transmitting voice conversations over the internet has been available to end users since at least the 1990's. For instance, in 1996, a shrink-wrapped software product called Vocaltec Internet Phone Release 4 provided VoIP, along with extra features such as voice mail and caller id. However, it did not offer a gateway to the analog POTS system, so it was only possible to speak to other Vocaltec Internet Phone users.^[2] VocalTec is significant for their breakthroughs in realtime voice compression,^[3] which was vital at a time when the majority of users had at most a 28.8 kb/s dialup modem. In 1997, Level 3 began development of its first softswitch (a term they invented in 1998); softswitches were designed to replace a traditional hardware switchboards by serving as the gateway between two telephone networks.^[4] Voicemail (or voice mail; abbreviated v-mail or vmail) is a specific application of an interactive voice response system. ... For the protein involved in the synthesis of major histocompatibility complex II, see CLIP (protein). ... In computer science, real-time computing (RTC) is the study of hardware and software systems which are subject to a real-time constraintâ€”i. ... Voice compression may mean different things: Speech encoding refers to compression for transmission or storage, possibly to an unintelligible state, with decompression used prior to playback. ... Level 3 is an international communications and information services company and is headquartered in Broomfield, Colorado. ... A softswitch is a central device in a telephone network which connects calls from one phone line to another, entirely by means of software running on a computer system. ... The term switchboard, when used by itself can refer to: Telephone switchboard Electric switchboard This is a disambiguation page — a navigational aid which lists other pages that might otherwise share the same title. ...

Functionality

VoIP can facilitate tasks that may be more difficult to achieve using traditional networks that have been typically used historically:

A telephone company (or telco) provides telecommunications services such as telephony and data communications. ... The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications. ... A Secure Terminal Equipment desk set. ... This article is about algorithms for encryption and decryption. ... This page deals with authentication in computing. ...

Implementation

Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (known as QoS) guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round trip propagation delay (400 milliseconds to 600 milliseconds for geostationary satellite). 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. User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. ... In the fields of packet-switched networks and computer networking, the traffic engineering term Quality of Service, abbreviated QoS, refers to resource reservation control mechanisms. ... Latency is a time delay between the moment something is initiated, and the moment one of its effects begins. ... In telecommunication, jitter is an abrupt and unwanted variation of one or more signal characteristics, such as the interval between successive pulses, the amplitude of successive cycles, or the frequency or phase of successive cycles. ...

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. This article is about the network security device. ... In computer networking, Network Address Translation (NAT, also known as Network Masquerading, Native Address Translation or IP Masquerading) is a technique of transceiving network traffic through a router that involves re-writing the source and/or destination IP addresses and usually also the TCP/UDP port numbers of IP packets... A Session Border Controller (SBC) is a device used in some VoIP networks to exert control over the signaling and media streams involved in setting up, conducting, and tearing down calls. ... Skype (IPA: , rhymes with type) is a software program created by the entrepreneurs Niklas ZennstrÃ¶m and Janus Friis. ... A symmetric NAT is a network address translation where all requests from the same internal IP address and port, to a specific destination IP address and port, are mapped to the same external IP address and port. ... This article is about the Internet protocol. ... The Interactive Connectivity Establishment (ICE) draft, developed by the IETFs MMUSIC working group, provides a mechanism for NAT traversal, using various techniques. ...

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.^{[citation needed]}

Fixed delays cannot be controlled but some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, Diffserv). DiffServ or differentiated services is a method of trying to guarantee quality of service on large networks such as the Internet. ...

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. Teletraffic engineering is the application of traffic engineering theory to telecommunications. ...

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. In telecommunication, jitter is an abrupt and unwanted variation of one or more signal characteristics, such as the interval between successive pulses, the amplitude of successive cycles, or the frequency or phase of successive cycles. ... In telecommunication, a buffer is a routine or storage medium used in telecommunications to compensate for a difference in rate of flow of data, or time of occurrence of events, when transferring data from one device to another. ... In computing, buffer underrun is a state occurring when a buffer used to communicate between two devices or processes is fed with data at a lower speed than the data is being read from it. ...

Common causes of echo include impedance mismatches in analog circuitry, and acoustic coupling of the transmit and receive signal at the receiving end. Impedance mismatch has two meanings. ...

Reliability

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 in the absence of a uninterruptible power supply or generator. 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. A telephone line (or just line) is a single-user circuit on a telephone communications system. ... A power outage is the loss of the electricity supply to an area. ... This article is about machines that produce electricity. ... A telephone operator manually connecting calls with patch cables at a telephone switchboard. ... An uninterruptible power supply (UPS), also known as an uninterruptible power source or a battery backup is a device which maintains a continuous supply of electric power to connected equipment by supplying power from a separate source when utility power is not available. ... PBX redirects here. ... A modern GE cordless telephone, model 26930 A cordless telephone or portable telephone is a telephone with a wireless handset which communicates via radio waves with a base station connected to a fixed telephone line (POTS) and can only be operated near (typically less than 100 meters) its base station...

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 are 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 more reliable. In networking and in graph theory, capillary routing, for a given network, is a multi-path solution between a pair of source and destination nodes. ... Fountain codes are a class of Erasure codes with the property that a potentially limitless sequence of encoding symbols can be generated from a given set of source symbols such that the original source symbols can be recovered from any subset of the encoding symbols of size equal to or... In computer science, raptor codes are one of the first known classes of fountain codes with linear time encoding and decoding. ...

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. [info from searchnetworking. ...

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 fax image. A codec is a device or program capable of performing encoding and decoding on a digital data stream or signal. ... T.38 is an ITU recommendation for sending fax messages over IP networks in real time by encapsulating a standard T.30 fax data stream. ... For other uses, see Fax (disambiguation). ... The Internet Printing Protocol or IPP, defines a standard protocol for printing as well as managing print jobs, media size, resolution, and so forth. ...

Emergency calls

The nature of IP makes it difficult to locate network users geographically. Emergency calls, therefore, cannot easily be routed to a nearby call center, 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.^[5] The Internet Protocol (IP) is a data-oriented protocol used for communicating data across a packet-switched internetwork. ... Many countries public telephone networks have a single emergency telephone number, sometimes known as the universal emergency telephone number or occasionally the emergency services number, that allows a caller to contact local emergency services for assistance. ...

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.^{[citation needed]} 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.^{[citation needed]} A mobile phone operator (also wireless carrier) is a telephone company that provides phone services for mobile phone subscribers. ...

Integration into global telephone number system

While the traditional Plain Old Telephone Service (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. Plain old telephone service, or POTS, are the services available from analogue telephones prior to the introduction of electronic telephone exchanges into the public switched telephone network. ... E.164 is an ITU-T recommendation which defines the international public telecommunication numbering plan used in the PSTN and some other data networks. ...

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.^[6] However, note that many hotspots require browser-based authentication, which most SIP phones do not support.^[7] Wi-Fi (or Wi-fi, WiFi, Wifi, wifi), short for Wireless Fidelity, is a set of standards for wireless local area networks (WLAN) currently based on the IEEE 802. ... Hotspots are venues that offer Wi-Fi access. ...

Mobile phones & Hand held Devices

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.^[8] A telephone company (or telco) provides telecommunications services such as telephony and data communications. ... Market penetration is one of the four growth strategies as defined by Ansoff. ... While the term wireless network may technically be used to refer to any type of network that is wireless, the term is most commonly used to refer to a telecommunications network whose interconnections between nodes is implemented without the use of wires, such as a computer network (which is a... VoWLAN (Voice over Wireless LAN) is the use of a wireless broadband network for the purpose of vocal conversation. ... 3G is the third generation of mobile phone standards and technology, after 2G. It is based on the International Telecommunication Union (ITU) family of standards under the International Mobile Telecommunications programme, IMT-2000. 3G technologies enable network operators to offer users a wider range of more advanced services while achieving...

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 hand held devices using VOIP, the nicknames of "MoIP" and MVoip (Mobile VoIP) have been attributed to these mobile applications. This article is about the telecommunications corporation. ... MoIP, or mobile communications over internet protocol, is the mobilization of peer-to-peer communications including chat and talk using internet protocol via standard mobile communications applications including 3G, GPRS and Wifi. ... Mobile VoIP or Mobile Voice over Internet Protocol is the application of voice over IP technology to mobile handsets. ...

Hand held 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 hand held devices use WiFi of some sort. Global System for Mobile communications (GSM: originally from Groupe SpÃ©cial Mobile) is the most popular standard for mobile phones in the world. ... General Packet Radio Service (GPRS) is a mobile data service available to users of GSM mobile phones. ...

Another addition to hand held devices are ruggedized bar code 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 communications.

Security

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.^[9] 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. In cryptography and computer security, security through obscurity (sometimes security by obscurity) is to some a controversial principle in security engineering, which attempts to use secrecy (of design, implementation, etc. ... To eavesdrop is to surreptitiously overhear a private conversation. ... The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications. ... ZRTP is an extension to Real-time Transport Protocol (RTP) which describes a method of Diffie-Hellman key agreement for Secure Real-time Transport Protocol (SRTP). ... An Analog Telephony Adapter, or ATA, is a device used to connect one or more standard analog telephones to a Voice over IP based network. ... In computing, a soft phone is software that simulates a real phone and runs on a general purpose computer, rather than a dedicated device. ... IPsec (IP security) is a suite of protocols for securing Internet Protocol (IP) communications by authenticating and/or encrypting each IP packet in a data stream. ... This article does not cite any references or sources. ... Opportunistic Encryption (OE) allows for encryption for secure communication without any pre-arrangement specific to the pair of systems involved. ... Skype (IPA: , rhymes with type) is a software program created by the entrepreneurs Niklas ZennstrÃ¶m and Janus Friis. ... The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications. ...

The Voice VPN solution provides secure voice for enterprise VoIP networks by applying IPSec encryption to the digitized voice stream. To meet Wikipedias quality standards, this article or section may require cleanup. ... Secure voice (alternatively secure speech or ciphony) is a term in cryptography for devices which are designed to provide voice encryption for voice communication over a range of communication types such as radio, telephone or IP. A digital secure voice usually includes two components, a digitizer to convert between speech... IPsec (IP security) is a suite of protocols for securing Internet Protocol (IP) communications by authenticating and/or encrypting each IP packet in a data stream. ...

Pre-Paid Phone Cards

VoIP has become an important technology for phone services to travelers, migrant workers and expatriates, 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.

Caller ID

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. For the protein involved in the synthesis of major histocompatibility complex II, see CLIP (protein). ...

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. In the context of network security, a spoofing attack is a situation in which one person or program successfully masquerades as another by falsifying data and thereby gaining an illegitimate advantage. ...

VoIM

Voice over Instant Messaging (VoIM) presents VoIP as one communication mode among several, with an IM user interface (contact list and presence) as the primary user experience. Many instant messenger services added client-to-client or client-to-PSTN VoIP in the mid-2000s. // Instant messaging (IM) is a form of real-time communication between two or more people based on typed text. ...

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. 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. This is a list of commercial voice over IP network providers, arranged alphabetically with no restriction to region. ... A WildBlue Satellite Internet dish. ... The public switched telephone network (PSTN) is the network of the worlds public circuit-switched telephone networks, in much the same way that the Internet is the network of the worlds public IP-based packet-switched networks. ... Direct Inward Dialing (DID, also called DDI in Europe) is a feature offered by telephone companies for use with their customers PBX system, whereby the telephone company (telco) allocates a range of numbers all connected to their customers PBX. As calls are presented to the PBX, the number that... For other uses, see Europe (disambiguation). ... For other uses, see Asia (disambiguation). ...

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. Plain old telephone service, or POTS, are the services available from analogue telephones prior to the introduction of electronic telephone exchanges into the public switched telephone network. ... In computing, a soft phone is software that simulates a real phone and runs on a general purpose computer, rather than a dedicated device. ...

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. Gizmo Project is the name of a peer-to-peer VoIP network and of a proprietary freeware soft phone for that network. ... Skype (IPA: , rhymes with type) is a software program created by the entrepreneurs Niklas ZennstrÃ¶m and Janus Friis. ... The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. ... A telephone numbering plan is a system that allows subscribers to make and receive telephone calls across long distances. ...

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. For other uses, see 911 (disambiguation). ... Emergency services are public services that deal with emergencies and other aspects of Public Safety. ... A dispatch can be: A report sent to a newspaper by a correspondent. ... For other uses, see 911 (disambiguation). ...

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. For other uses, see Fax (disambiguation). ... TiVo (pronounced tee-voh, IPA: ) is a popular brand of digital video recorder (DVR) in the United States (and coming to Canada in December 7, 2007) and is a consumer video device which allows users to capture television programming to internal hard disk storage for later viewing (time shifting), provides... ReplayTV is a brand of digital video recorder (DVR), a term synonymous with personal video recorder (PVR). ... Satellite television is television delivered by way of communications satellites, as compared to conventional terrestrial television and cable television. ... For other uses, see Alarm (disambiguation). ... For other uses, see Modem (disambiguation). ... For other uses, see Telephone (disambiguation). ... In telecommunications and computing, bit rate (sometimes written bitrate) is the frequency at which bits are passing a given (physical or metaphorical) point. It is quantified using the bit per second (bit/s) unit. ... The tone or style of this article or section may not be appropriate for Wikipedia. ... TestYourVoIP is a free web service run by Brix Networks that allows you to test your internet connection to see how well it will perform when used for VoIP calls. ... A Java Virtual Machine (JVM) is a set of computer software programs and data structures which implements a specific virtual machine model. ...

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. The IP Multimedia Subsystem (IMS) is an architectural framework for delivering internet protocol (IP) multimedia to mobile users. ...

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. Virtual PBX (or IP PBX) allow companies to control their internal phone network over an existing LAN and server without needing to wire a separate telephone network. Users within this environment can then use standard telephones coupled with an FXS, IP Phones connected to a data port or a Softphone on their PC. Internal VoIP phone networks allow outbound and inbound calling on standard PSTN lines through the use of FXO adapters. TElephone NUMber Mapping (ENUM or Enum) is a suite of protocols to unify the telephone system with the Internet by using E.164 addresses with DDDS and DNS. The acronymn ENUM can also mean E164 NUmber Mapping. ... E.164 is an ITU-T recommendation which defines the international public telecommunication numbering plan used in the PSTN and some other data networks. ... The Internet Protocol Private Branch eXchange (IP PBX) is telephone switching equipment that resides in a private business instead of the telephone company. ... Lan can stand for several things: A local area network Lan (airline) formerly LanChile Lan Peru LÃ¤n, a kind of administrative division used in Sweden Lan Mandragoran, a fictional character in the Wheel of Time fantasy series by Robert Jordan. ... FXS can stand for: foreign exchange station fragile X syndrome This is a disambiguation page â€” a list of articles associated with the same title. ... In computing, a soft phone is software that simulates a real phone and runs on a general purpose computer, rather than a dedicated device. ... A Foreign Exchange Office, or FXO, is a telephone interface that receives POTS, or Plain old telephone service. ...

Use in Amateur Radio

Sometimes called Radio Over Internet Protocol or RoIP, Amateur radio has adopted VoIP by linking repeaters and users with Echolink, IRLP, D-STAR, Dingotel and EQSO. 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. This page is a candidate to be copied to Wiktionary. ... Echolink is a computer program that runs under Microsoft Windows to allow radio amateurs to communicate with one another using Voice over IP (VoIP) technology on the internet for at least part of the path between them. ... The Internet Radio Linking Project is a project within general amateur radio. ... D-STAR (Digital Smart Technologies for Amateur Radio) is a digital voice and data protocol specification developed for use in amateur radio. ... eQSO is a client-server software program designed by amateur radio enthusiasts for linking amateur radio frequency gateways and repeaters via the internet. ... Echolink is a computer program that runs under Microsoft Windows to allow radio amateurs to communicate with one another using Voice over IP (VoIP) technology on the internet for at least part of the path between them. ...

Ham Radio operators using radios are able to tune to repeaters with VoIP capabilities and use DTMF signals to command the repeater to connect to various other repeaters, thus allowing them to talk to people all around the world, even with "line of sight" VHF radios. Dual-tone multi-frequency (DTMF), also known as Touch Tone® is used for telephone signaling over the line in the voice frequency band to the call switching center. ... Very high frequency (VHF) is the radio frequency range from 30 MHz to 300 MHz. ...

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. Click-To-Call or CTC is a service which lets users click a button and immediately speak with a customer service representative or other user. ...

Legal issues in different countries

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,^[10] 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 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. FCC redirects here. ... To meet Wikipedias quality standards and make it easier to understand, this article or section may require cleanup. ... The Communications Assistance for Law Enforcement Act (CALEA) is a controversial United States wiretapping law passed in 1994 (Pub. ... The Telecommunications Act of 2005 (tenatively named) is a proposed United States telecommunications law that makes regulatory changes to broadband Internet providers, Voice over IP providers, and Broadband Video services. ... LNP redirects here. ... In telecommunications, universal service was conceived by Theodore Vail, at AT&T, in the late 1800s; any user could connect. ... Telephone tapping (or wire tapping/wiretapping in the US) is the monitoring of telephone and Internet conversations by a third party, often by covert means. ... Network neutrality (equivalently net neutrality, Internet neutrality or NN) refers to a principle that is applied to residential broadband networks, and potentially to all networks. ...

Some Latin American and Caribbean 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 the government is monopolizing telecommunication service, it is a criminal offense 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. Latin America consists of the countries of South America and some of North America (including Central America and some the islands of the Caribbean) whose inhabitants mostly speak Romance languages, although Native American languages are also spoken. ... West Indies redirects here. ...

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 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 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 should not be inside India. POTS may mean: Plain old telephone service (aka Public Switched Telephone Network (PSTN) or Post Office Telephone Service or Post Office Telephone System) Postural orthostatic tachycardia syndrome This article consisting of a 4-letter acronym or initialism is a disambiguation page â€” a list of pages that otherwise might share the...

In the UAE, it is illegal to use any form of VoIP, to the extent that websites of Skype and Gizmo Project don't work. UAE redirects here; for other uses of that term, see UAE (disambiguation) The United Arab Emirates is an oil-rich country situated in the south-east of the Arabian Peninsula in Southwest Asia, comprising seven emirates: Abu Dhabi, Ajman, Dubai, Fujairah, Ras al-Khaimah, Sharjah and Umm al-Quwain. ... Skype (IPA: , rhymes with type) is a software program created by the entrepreneurs Niklas ZennstrÃ¶m and Janus Friis. ... Gizmo Project is the name of a peer-to-peer VoIP network and of a proprietary freeware soft phone for that network. ...

In the Republic of 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 such as Vonage 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 of 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 U.S.-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 U.S. VoIP providers.^[11] Vonage Logo until 2006 Vonage (NYSE: VG) (pronounced ) is a publicly held commercial voice over IP (VoIP) network and SIP company that provides telephone service via a broadband connection (the companys name is a play on their motto Voice-Over-Net-AGE). Vonage promotes itself as Vonage the Broadband... Year 2006 (MMVI) was a common year starting on Sunday of the Gregorian calendar. ... An Internet Service Provider (ISP) is a business or organization that offers users access to the Internet and related services. ... United States Forces Korea (USFK, Korean: ì£¼í•œë¯¸êµ°, Hanja: é§éŸ“ç¾Žè») refers to the ground, air and naval divisions of the United States Armed Forces stationed in South Korea. ... January 2007 is the first month of that year. ...

IP telephony in Japan

In Japan, IP telephony (IP電話 IP Denwa ^?) is regarded as a service applied VoIP technology to whole or a part of the telephone line. As from 2003, IP telephony service assigned telephone numbers has been provided. There are not voice only services, but also videophone service. According to the Telecommunication Business Law, the service category for IP telephony also implies the service provided via Internet, which is not assigned any telephone number. IP telephony is basically regulated by Ministry of Internal Affairs and Communications (MIC), as a telecommunication service. The operators have to disclose necessary information on its quality, etc, prior to making contract with customers, and have obligation to respond to their complaints cordially. Ministry of Internal Affairs and Communications (総務省 Soumu-sho) is one of ministries in the Cabinet of Japan. ...

Many Internet service providers (ISP) are providing IP telephony services. The provider, which provides IP telephony service, is so-called "ITSP (Internet Telephony Service Provider)". Recently, the competition among ITSPs has been activated, by option or set sales, connected with ADSL or FTTH services.

The tariff system normally applied for Japanese IP telephony tends to be described as below;

Between ITSP, the interconnection is mostly maintained at VoIP level. The Public Health Service Commissioned Corps is the uniformed division of the United States Public Health Service (PHS) and one of the seven Uniformed Services of the United States. ...

Telephone number for IP telephony in Japan

Since September 2002, the MIC has assigned IP telephony telephone numbers on the condition that the service falls into certain required categories of quality. Highly qualified IP telephony is assigned a telephone number. Normally the number starts with 050. But, when its quality is so high that customer almost could not tell the difference between it and a normal telephone and when the provider relates its number with a location and provides the connection with emergency call capabilities, the provider is allowed to assign a normal telephone number, which is a so-called "0AB-J" number. 2002 : January - February - March - April - May - June - July - August - September - October - November - December A timeline of events in the news for September, 2002. ...

Technical details

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 in the "local loop" it has since been surpassed by SIP. This was primarily due to the latter's better traversal of NAT and firewalls, although recent changes introduced for H.323 have removed this advantage.^{[citation needed]} The Internet Engineering Task Force (IETF) is charged with developing and promoting Internet standards. ... The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. ... This article is about the location. ... H.323 is an umbrella recommendation from the ITU-T, that defines the protocols to provide audio-visual communication sessions on any packet network. ...

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^{[citation needed]}, and the vast majority of callers have little or no idea that their POTS calls are being carried over VoIP.

Where VoIP travels through multiple providers' softswitches the concepts 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 Signaling; 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 signaling proxy mode where only the signaling flows through the provider the delay will be reduced to a more user friendly 120-150 ms. A softswitch is a central device in a telephone network which connects calls from one phone line to another, entirely by means of software running on a computer system. ... H.225. ... H.245 is the Control Protocol for Multimedia Communication within H.323. ...

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 requires 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 optimization techniques used, such as silence suppression and header compression. This can typically save 35% on bandwidth usage. G.723. ...

VoIP trunking techniques such as TDMoIP can reduce bandwidth overhead even further by multiplexing multiple conversations that are heading to the same destination and wrapping them up inside the same packets. Because the packet header overhead is shared between many simultaneous streams, TDMoIP can offer near toll quality audio with a per-stream packet header overhead of only about 1 kbit/s. In computer networking and telecommunications, TDM over IP (TDMoIP) is the emulation of time-division multiplexing (TDM) over a packet switched network (PSN). ...

Contents

History

Functionality

Implementation

Reliability

Quality of service

Difficulty with sending faxes

Emergency calls

Integration into global telephone number system

Single point of calling

Mobile phones & Hand held Devices

Security

Pre-Paid Phone Cards

Caller ID

VoIM

Adoption

Mass-market telephony

Corporate and telco use

Use in Amateur Radio

Click to call

Legal issues in different countries

IP telephony in Japan

Telephone number for IP telephony in Japan

Technical details

See also

References

External links

Contents

History GA_googleFillSlot("encyclopedia_square");

Functionality

Implementation

Reliability

Quality of service

Difficulty with sending faxes

Emergency calls

Integration into global telephone number system

Single point of calling

Mobile phones & Hand held Devices

Security

Pre-Paid Phone Cards

Caller ID

VoIM

Adoption

Mass-market telephony

Corporate and telco use

Use in Amateur Radio

Click to call

Legal issues in different countries

IP telephony in Japan

Telephone number for IP telephony in Japan

Technical details

See also

References

External links

History