voice over ip

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Voice over IPFrom Wikipedia, the free encyclopedia

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Voice-over-Internet Protocol(VoIP) is a methodology and group of technologies for the delivery

ofvoice communicationsandmultimediasessions overInternet Protocol(IP) networks, such as theInternet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, voiceover broadband(VoBB), broadband telephony, IP communications, and broadband phone service.

The term Internet telephonyspecifically refers to the provisioning of communications services(voice,fax,SMS, voice-messaging) over the publicInternet,rather than via thepublic switchedtelephone network(PSTN). The steps and principles involved in originating VoIP telephone calls aresimilar to traditional digitaltelephonyand involve signaling, channel setup, digitization of the analogvoice signals, and encoding. Instead of being transmitted over a circuit-switched network, however,the digital information is packetized, and transmission occurs as IP packets over a packet-switchednetwork.Such transmission entails careful considerations about resource management differentfromtime-division multiplexing(TDM) networks.

Early providers of voice-over-IP services offered business models and technical solutions that

mirrored the architecture of the legacy telephone network. Second-generation providers, suchasSkype,have built closed networks for private user bases, offering the benefit of free calls andconvenience while potentially charging for access to other communication networks, such as thePSTN. This has limited the freedom of users to mix-and-match third-party hardware and software.Third-generation providers, such asGoogle Talk,have adopted[1]the concept offederated VoIPwhich is a departure from the architecture of the legacy networks. These solutions typically allowdynamic interconnection between users on any two domains on the Internet when a user wishes toplace a call.

VoIP systems employ session control and signaling protocols to control the signaling, set-up, andtear-down of calls. They transport audio streams over IP networks using special media deliveryprotocols that encode voice, audio, video withaudio codecs,and video codecs asDigitalaudiobystreaming media.Various codecs exist that optimize the media stream based on

application requirements and network bandwidth; some implementations relyonnarrowbandandcompressed speech,while others supporthigh fidelitystereo codecs. Somepopular codecs include-lawanda-lawversions ofG.711,G.722,which is a high-fidelity codecmarketed as HD Voice byPolycom,a popular open source voice codec known asiLBC,a codec thatonly uses 8 kbit/s each way calledG.729,and many others.

VoIP is available on manysmartphones,personal computers, and on Internet access devices. Callsand SMS text messages may be sent over3GorWi-Fi.[2]

Contents

[hide]

1 Pronunciation

2 Protocols

3 Adoption

o 3.1 Consumer market

o 3.2 PSTN and mobile network providers

o 3.3 Corporate use

4 Quality of service

o 4.1 Layer 2

5 PSTN integration
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o 5.1 Number portability

o 5.2 Emergency calls

6 Fax support

7 Power requirements

8 Redundancy

9 Security

10 Caller ID

11 Compatibility with traditional analog telephone sets

12 Support for other telephony devices

13 User and administrative interfaces

14 Operational cost

15 Regulatory and legal issues

o 15.1 European Union

o 15.2 Pakistan

o 15.3 India

o 15.4 Middle East

o 15.5 South Korea

o

15.6 United States 16 Historical milestones

17 See also

18 References

19 External links

Pronunciation[edit]

The acronym "VoIP" has been pronounced variably since the inception of the term. Apart fromspelling out the acronym letter by letter, /vioapi/ (vee-oh-eye-pee), there are three likely possiblepronunciations: /voapi/ (vo-eye-pee) and /voip/ (vo-ipp), have been used, but generally, thesingle syllable /vjp/ (voyp, as in voice) may be the most common within the industry.[3]

Protocols[edit]

Voice over IP has been implemented in various ways using bothproprietary protocolsand protocolsbased onopen standards.Examples of the VoIP protocols are:

H.323

Media Gateway Control Protocol (MGCP)

Session Initiation Protocol(SIP)

H.248(also known as Media Gateway Control (Megaco))

Real-time Transport Protocol(RTP)

Real-time Transport Control Protocol(RTCP)

Secure Real-time Transport Protocol(SRTP) Session Description Protocol(SDP)

Inter-Asterisk eXchange(IAX)

JingleXMPPVoIP extensions

Skype protocol

Teamspeak

The H.323 protocol was one of the first VoIP protocols that found widespread implementation forlong-distance traffic, as well aslocal area networkservices. However, since the development of
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scription_Protocolhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Real-time_Transport_Control_Protocolhttp://en.wikipedia.org/wiki/Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/H.248http://en.wikipedia.org/wiki/Session_Initiation_Protocolhttp://en.wikipedia.org/wiki/Media_Gateway_Control_Protocol_(MGCP)http://en.wikipedia.org/wiki/H.323http://en.wikipedia.org/wiki/Open_Standard#Protocolshttp://en.wikipedia.org/wiki/Proprietary_protocolhttp://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=2http://en.wikipedia.org/wiki/Voip#cite_note-3http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=1http://en.wikipedia.org/wiki/Voip#External_linkshttp://en.wikipedia.org/wiki/Voip#Referenceshttp://en.wikipedia.org/wiki/Voip#See_alsohttp://en.wikipedia.org/wiki/Voip#Historical_milestoneshttp://en.wikipedia.org/wiki/Voip#United_Stateshttp://en.wikipedia.org/wiki/Voip#South_Koreahttp://en.wikipedia.org/wiki/Voip#Middle_Easthttp://en.wikipedia.org/wiki/Voip#Indiahttp://en.wikipedia.org/wiki/Voip#Pakistanhttp://en.wikipedia.org/wiki/Voip#European_Unionhttp://en.wikipedia.org/wiki/Voip#Regulatory_and_legal_issueshttp://en.wikipedia.org/wiki/Voip#Operational_costhttp://en.wikipedia.org/wiki/Voip#User_and_administrative_interfaceshttp://en.wikipedia.org/wiki/Voip#Support_for_other_telephony_deviceshttp://en.wikipedia.org/wiki/Voip#Compatibility_with_traditional_analog_telephone_setshttp://en.wikipedia.org/wiki/Voip#Caller_IDhttp://en.wikipedia.org/wiki/Voip#Securityhttp://en.wikipedia.org/wiki/Voip#Redundancyhttp://en.wikipedia.org/wiki/Voip#Power_requirementshttp://en.wikipedia.org/wiki/Voip#Fax_supporthttp://en.wikipedia.org/wiki/Voip#Emergency_callshttp://en.wikipedia.org/wiki/Voip#Number_portability


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newer, less complex protocols such as MGCP and SIP, H.323 deployments are increasingly limitedto carrying existing long-haul network traffic. In particular, the Session Initiation Protocol (SIP) hasgained widespread VoIP market penetration.

These protocols can be used by special-purpose software, such asJitsi,or integrated into a webpage (web-based VoIP), likeGoogle Talk.

Adoption[edit]Consumer market[edit]

Example of residential network including VoIP

A major development that started in 2004 was the introduction of mass-market VoIP services thatutilize existingbroadband Internet access,by which subscribers place and receive telephone calls inmuch the same manner as they would via thepublic switched telephone network(PSTN). Full-service VoIP phone companies provide inbound and outbound service withdirect inbound dialing.Many offer unlimited domestic calling for a flat monthly subscription fee. This sometimes includes

international calls to certain countries. Phone calls between subscribers of the same provider areusually free when flat-fee service is not available. AVoIP phoneis necessary to connect to a VoIPservice provider. This can be implemented in several ways:

Dedicated VoIP phones connect directly to the IP network using technologies such aswiredEthernetorWi-Fi.They are typically designed in the style of traditional digital businesstelephones.

Ananalog telephone adapteris a device that connects to the network and implements theelectronics and firmware to operate a conventional analog telephone attached through amodular phone jack. Some residential Internet gateways andcablemodemshave this functionbuilt in.

Asoftphoneis application software installed on a networked computer that is equipped with a

microphone and speaker, or headset. The application typically presents a dial pad and displayfield to the user to operate the application by mouse clicks or keyboard input.

PSTN and mobile network providers[edit]

It is becoming increasingly common for telecommunications providers to use VoIP telephony overdedicated and public IP networks to connect switching centers and to interconnect with othertelephony network providers; this is often referred to as "IPbackhaul."[4][5]

SmartphonesandWi-Fi-enabled mobile phones may have SIP clients built into the firmware oravailable as an application download.
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Corporate use[edit]

Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses aremigrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthlyphone costs. In 2008, 80% of all newPrivate branch exchange(PBX) lines installed internationallywere VoIP.[6]

VoIP solutions aimed at businesses have evolved intounified communicationsservices that treat allcommunicationsphone calls, faxes, voice mail, e-mail, Web conferences, and moreas discreteunits that can all be delivered via any means and to any handset, including cellphones. Two kinds ofcompetitors are competing in this space: one set is focused on VoIP for medium to large enterprises,while another is targeting the small-to-medium business (SMB) market.[7]

VoIP allows both voice and data communications to be run over a single network, which cansignificantly reduce infrastructure costs.[8]

The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may runon commodity hardware, such aspersonal computers.Rather than closed architectures, thesedevices rely on standard interfaces.[8]

VoIP devices have simple, intuitive user interfaces, so users can often make simple system

configuration changes. Dual-mode phones enable users to continue their conversations as theymove between an outside cellular service and an internalWi-Finetwork, so that it is no longernecessary to carry both a desktop phone and a cellphone. Maintenance becomes simpler as thereare fewer devices to oversee.[8]

Skype,which originally marketed itself as a service among friends, has begun to cater tobusinesses, providing free-of-charge connections between any users on the Skype network andconnecting to and from ordinaryPSTNtelephones for a charge.[9]

In the United States the Social Security Administration (SSA) is converting its field offices of 63,000workers from traditional phone installations to a VoIP infrastructure carried over its existing datanetwork.[10][11]

Quality of service[edit]Communication on the IP network is perceived as less reliable in contrast to the circuit-switchedpublic telephone network because it does not provide a network-based mechanism to ensure thatdata packets are not lost, and are delivered in sequential order. [citation needed]It is a best-effort networkwithout fundamentalQuality of Service(QoS) guarantees. Therefore, VoIP implementations mayface problems withlatency,packet loss, andjitter.[12][13]

By default, network routers handle traffic on a first-come, first-served basis. Network routers on highvolume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Fixed delayscannot be controlled, as they are caused by the physical distance the packets travel; however,latency can be minimized by marking voice packets as being delay-sensitive with methods suchasDiffServ.[12]

VoIP endpoints usually have to wait for completion of transmission of previous packets before newdata may be sent. Although it is possible to preempt (abort) a less important packet in mid-transmission, this is not commonly done, especially on high-speed links where transmission timesare short even for maximum-sized packets.[14]An alternative to preemption on slower links, such asdialup anddigital subscriber line(DSL), is to reduce the maximum transmission time by reducingthemaximum transmission unit.But every packet must contain protocol headers, so this increasesrelative header overhead on every link traversed, not just the bottleneck (usually Internet access)link.[14]
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ttp://en.wikipedia.org/wiki/Unified_communicationshttp://en.wikipedia.org/wiki/Voip#cite_note-6http://en.wikipedia.org/wiki/Private_branch_exchangehttp://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=6


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DSL modems provide Ethernet (or Ethernet overUSB)connections to local equipment, but insidethey are actuallyAsynchronous Transfer Mode(ATM) modems. They useATM Adaptation Layer5(AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission,reassembling them back into Ethernet frames at the receiving end. Avirtual circuit identifier(VCI) ispart of the 5-byte header on every ATM cell, so the transmitter canmultiplexthe active virtual circuits(VCs) in any arbitrary order. Cells from the sameVC are always sent sequentially.

However, a majority of DSL providers use only one VC for each customer, even those with bundledVoIP service. Every Ethernet frame must be completely transmitted before another can begin. If asecond VC were established, given high priority and reserved for VoIP, then a low priority datapacket could be suspended in mid-transmission and a VoIP packet sent right away on the highpriority VC. Then the link would pick up the low priority VC where it left off. Because ATM links aremultiplexed on a cell-by-cell basis, a high priority packet would have to wait at most 53 byte times tobegin transmission. There would be no need to reduce the interface MTU and accept the resultingincrease in higher layer protocol overhead, and no need to abort a low priority packet and resend itlater.

ATM has substantial header overhead: 5/53 = 9.4%, roughly twice the total header overhead of a1500 byte Ethernet frame. This "ATM tax" is incurred by every DSL user whether or not they takeadvantage of multiple virtual circuits - and few can.[12]

ATM's potential for latency reduction is greatest on slow links, because worst-case latencydecreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmitat 128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link, this latency is probably smallenough to ensure good VoIP performance without MTU reductions or multiple ATM VCs. The latestgenerations of DSL,VDSLandVDSL2,carry Ethernet without intermediate ATM/AAL5 layers, andthey generally supportIEEE 802.1ppriority tagging so that VoIP can be queued ahead of less time-critical traffic.[12]

Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. Thissystem may be more prone to congestion[citation needed]andDoS attacks[15]than traditionalcircuitswitchedsystems; a circuit switched system of insufficient capacity will refuse new connections whilecarrying the remainder without impairment, while the quality of real-time data such as telephone

conversations on packet-switched networks degrades dramatically.

[12]

Fixed delays cannot be controlled as they are caused by the physical distance the packets travel.They are especially problematic when satellite circuits are involved because of the long distance toageostationary satelliteand back; delays of 400600 ms are typical.

When the load on a link grows so quickly that its switches experience queue overflows, congestionresults and data packets are lost. This signals a transport protocol likeTCPto reduce itstransmission rate to alleviate the congestion. But VoIP usually usesUDPnot TCP becauserecovering from congestion through retransmission usually entails too much latency.[12]So QoSmechanisms can avoid the undesirable loss of VoIP packets by immediately transmitting themahead of any queued bulk traffic on the same link, even when that bulk traffic queue is overflowing.

The receiver must resequence IP packets that arrive out of order and recover gracefully whenpackets arrive too late or not at all.Jitterresults from the rapid and random (i.e. unpredictable)changes in queue lengths along a given Internet path due to competition from other users for thesame transmission links. VoIP receivers counter jitter by storing incoming packets briefly in a "de-

jitter" or "playout"buffer,deliberately increasing latency to improve the chance that each packet willbe on hand when it is time for thevoice engineto play it. The added delay is thus a compromisebetween excessive latency and excessivedropout,i.e. momentary audio interruptions.

Although jitter is a random variable, it is the sum of several other random variables that are at leastsomewhat independent: the individual queuing delays of the routers along the Internet path inquestion. Thus according to thecentral limit theorem,we can model jitter as agaussian random
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variable.This suggests continually estimating the mean delay and its standard deviation and settingthe playout delay so that only packets delayed more than several standard deviations above themean will arrive too late to be useful. In practice, however, the variance in latency of many Internetpaths is dominated by a small number (often one) of relatively slow and congested "bottleneck" links.Most Internet backbone links are now so fast (e.g. 10 Gbit/s) that their delays are dominated by thetransmission medium (e.g. optical fiber) and the routers driving them do not have enough buffering

for queuing delays to be significant.It has been suggested to rely on the packetized nature of media in VoIP communications andtransmit the stream of packets from the source phone to the destination phone simultaneouslyacross different routes (multi-path routing).[16]In such a way, temporary failures have less impact onthe communication quality. Incapillary routingit has been suggested to use at the packetlevelFountain codesor particularlyraptor codesfor transmitting extra redundant packets making thecommunication more reliable.[citation needed]

A number of protocols have been defined to support the reporting ofquality of service(QoS)andquality of experience(QoE) for VoIP calls. These includeRTCPExtended Report (RFC3611),SIPRTCP Summary Reports, H.460.9 Annex B (forH.323),H.248.30andMGCPextensions.The RFC 3611 VoIP Metrics block is generated by an IP phone or gateway during a live call andcontains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard

burst metrics (burst length/density, gap length/density), network delay, end system delay, signal /noise / echo level,Mean Opinion Scores(MOS) and R factors and configuration information relatedto the jitter buffer.

RFC 3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis duringa call, and an end of call message sent via SIP RTCP Summary Report or one of the other signalingprotocol extensions. RFC 3611 VoIP metrics reports are intended to support real time feedbackrelated to QoS problems, the exchange of information between the endpoints for improved callquality calculation and a variety of other applications.

Rural areas in particular are greatly hindered in their ability to choose a VoIP system over PBX. Thisis generally down to the poor access to superfast broadband in rural country areas. With the releaseof 4G data, there is a potential for corporate users based outside of populated areas to switch their

internet connection to 4G data, which is comparatively as fast as a regular superfast broadbandconnection. This greatly enhances the overall quality and user experience of a VoIP system in theseareas. This method was already trialled in rural Germany, surpassing all expectations.[17]

Layer 2[edit]

A number of protocols that deal with thedata link layerandphysical layerinclude quality-of-servicemechanisms that can be used to ensure that applications like VoIP work well even in congestedscenarios. Some examples include:

IEEE 802.11eis an approved amendment to theIEEE 802.11standard that defines a set ofquality-of-service enhancements for wireless LAN applications through modifications totheMedia Access Control(MAC) layer. The standard is considered of critical importance fordelay-sensitive applications, such as voice over wireless IP.

IEEE 802.1pdefines 8 different classes of service (including one dedicated to voice) for trafficon layer-2 wiredEthernet.

TheITU-TG.hnstandard, which provides a way to create a high-speed (up to 1 gigabit persecond)Local area network(LAN) using existing home wiring (power lines,phone linesandcoaxial cables). G.hn provides QoS by means of "Contention-Free TransmissionOpportunities" (CFTXOPs) which are allocated to flows (such as a VoIP call) which require QoSand which have negotiated a "contract" with the network controllers.
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iki/Quality_of_servicehttp://en.wikipedia.org/wiki/Wikipedia:Citation_neededhttp://en.wikipedia.org/wiki/Raptor_codehttp://en.wikipedia.org/wiki/Fountain_codehttp://en.wikipedia.org/wiki/Capillary_routinghttp://en.wikipedia.org/wiki/Voip#cite_note-16http://en.wikipedia.org/wiki/Gaussian_random_variable


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PSTN integration[edit]

The Media VoIP Gateway connects the digital media stream, so as to complete creating the path forvoice as well as data media. It includes the interface for connecting the standard PSTN networkswith the ATM and Inter Protocol networks. The Ethernet interfaces are also included in the modernsystems, which are specially designed to link calls that are passed via the VoIP.[18]

E.164is a global FGFnumbering standard for both thePSTNandPLMN.Most VoIPimplementations supportE.164to allow calls to be routed to and from VoIP subscribers and thePSTN/PLMN.[19]VoIP implementations can also allow other identification techniques to be used. Forexample, Skype allows subscribers to choose "Skype names"[20](usernames) whereas SIPimplementations can useURIs[21]similar toemail addresses.Often VoIP implementations employmethods of translating non-E.164 identifiers to E.164 numbers and vice-versa, such as the Skype-Inservice provided by Skype[22]and theENUMservice in IMS and SIP.[23]

Echo can also be an issue for PSTN integration.[24]Common causes of echo includeimpedancemismatchesin analog circuitry and acoustic coupling of the transmit and receive signal at thereceiving end.

Number portability[edit]

Local number portability(LNP) andMobile number portability(MNP) also impact VoIP business. InNovember 2007, theFederal Communications Commissionin the United States released an orderextending number portability obligations to interconnected VoIP providers and carriers that supportVoIP providers.[25]Number portability is a service that allows a subscriber to select a new telephonecarrier without requiring a new number to be issued. Typically, it is the responsibility of the formercarrier to "map" the old number to the undisclosed number assigned by the new carrier. This isachieved by maintaining a database of numbers. A dialed number is initially received by the originalcarrier and quickly rerouted to the new carrier. Multiple porting references must be maintained evenif the subscriber returns to the original carrier. The FCC mandates carrier compliance with theseconsumer-protection stipulations.

A voice call originating in the VoIP environment also faces challenges to reach its destination if thenumber is routed to a mobile phone number on a traditional mobile carrier. VoIP has been identifiedin the past as aLeast Cost Routing(LCR) system, which is based on checking the destination ofeach telephone call as it is made, and then sending the call via the network that will cost thecustomer the least. This rating is subject to some debate given the complexity of call routing createdby number portability. WithGSMnumber portability now in place, LCR providers can no longer relyon using the network root prefix to determine how to route a call. Instead, they must now determinethe actual network of every number before routing the call.

Therefore, VoIP solutions also need to handle MNP when routing a voice call. In countries without acentral database, like the UK, it might be necessary to query theGSMnetwork about which homenetwork a mobile phone number belongs to. As the popularity of VoIP increases in the enterprisemarkets because ofleast cost routingoptions, it needs to provide a certain level of reliability whenhandling calls.

MNP checks are important to assure that this quality of service is met. Handling MNP lookups beforerouting a call provides some assurance that the voice call will actually work.

Emergency calls[edit]

A telephone connected to aland linehas a direct relationship between a telephone number and aphysical location, which is maintained by the telephone company and available to emergencyresponders via the national emergency response service centers in form of emergency subscriberlists. When an emergency call is received by a center the location is automatically determined fromits databases and displayed on the operator console.
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In IP telephony, no such direct link between location and communications end point exists. Even aprovider having hardware infrastructure, such as a DSL provider, may only know the approximatelocation of the device, based on the IP address allocated to the network router and the knownservice address. However, some ISPs do not track the automatic assignment of IP addresses tocustomer equipment.[26]

IP communication provides for device mobility. For example, a residential broadband connectionmay be used as a link to avirtual private networkof a corporate entity, in which case the IP addressbeing used for customer communications may belong to the enterprise, not being the networkaddress of the residential ISP. Suchoff-premise extensionsmay appear as part of an upstream IPPBX. On mobile devices, e.g., a 3G handset or USB wireless broadband adapter, the IP addresshas no relationship with any physical location known to the telephony service provider, since amobile user could be anywhere in a region with network coverage, even roaming via another cellularcompany.

At the VoIP level, a phone or gateway may identify itself with aSession Initiation Protocol(SIP)registrar by its account credentials. In such cases, theInternet telephony service provider(ITSP) onlyknows that a particular user's equipment is active. Service providers often provide emergencyresponse services by agreement with the user who registers a physical location and agrees thatemergency services are only provided to that address if an emergency number is called from the IP

device.

Such emergency services are provided by VoIP vendors in the United States by a systemcalledEnhanced 911(E911), based on the Wireless Communications and Public Safety Act of 1999.The VoIP E911 emergency-calling system associates a physical address with the calling party'stelephone number. All VoIP providers that provide access to the public switched telephone networkare required to implement E911,[26]a service for which the subscriber may be charged. However,end-customer participation in E911 is not mandatory and customers may opt-out of the service .[26]

The VoIP E911 system is based on a static table lookup. Unlike in cellular phones, where thelocation of an E911 call can be traced usingassisted GPSor other methods, the VoIP E911information is only accurate so long as subscribers, who have the legal responsibility, are diligent inkeeping their emergency address information current.

Fax support[edit]

Support for fax has been problematic in many VoIP implementations, as most voice digitization andcompressioncodecsare optimized for the representation of the human voice and the proper timingof the modem signals cannot be guaranteed in a packet-based, connection-less network. Analternative IP-based solution for delivering fax-over-IP calledT.38is available. Sending faxes usingVoIP is sometimes referred to as FoIP, or Fax over IP.

The T.38 protocol is designed to compensate for the differences between traditional packet-lesscommunications over analog lines and packet-based transmissions which are the basis for IPcommunications. The fax machine could be a traditional fax machine connected to the PSTN, or an

ATA box (or similar). It could be a fax machine with an RJ-45 connector plugged straight into an IP

network, or it could be a computer pretending to be a fax machine .

[27]

Originally, T.38 was designedto use UDP and TCP transmission methods across an IP network. TCP is better suited for usebetween two IP devices. However, older fax machines, connected to an analog system, benefit fromUDP near real-time characteristics due to the "no recovery rule" when a UDP packet is lost or anerror occurs during transmission.[28]UDP transmissions are preferred as they do not require testingfor dropped packets and as such since each T.38 packet transmission includes a majority of the datasent in the prior packet, a T.38 termination point has a higher degree of success in re-assemblingthe fax transmission back into its original form for interpretation by the end device. This in an attemptto overcome the obstacles of simulating real time transmissions using packet based protocol.[29]
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There have been updated versions of T.30 to resolve the fax over IP issues, which is the core faxprotocol. Some newer high end fax machines have T.38 built-in capabilities which allow the user toplug right into the network and transmit/receive faxes in native T.38 like the Ricoh 4410NF FaxMachine.[30]A unique feature of T.38 is that each packet contains a portion of the main data sent inthe previous packet. With T.38, two successive lost packets are needed to actually lose any data.The data one will lose will only be a small piece, but with the right settings and error correction

mode, there is an increased likelihood that they will receive enough of the transmission to satisfy therequirements of the fax machine for output of the sent document.

While many late-modelanalog telephone adapters(ATAs) support T.38, uptake has been limited asmany voice-over-IP providers performleast-cost routingwhich selects the least expensive PSTNgateway in the called city for an outbound message. There is typically no means to ensure that thatgateway is T.38 capable. Providers often place their own equipment (such as anAsteriskPBXinstallation) in the signal path, which creates additional issues as every link in the chain mustbe T.38 aware for the protocol to work. Similar issues arise if a provider is purchasing local directinward dialnumbers from the lowest bidder in each city, as many of these may not be T.38 enabled.

Power requirements[edit]

Telephones for traditional residential analog service are usually connected directly to telephonecompanyphone lineswhich provide direct current to power most basic analog handsetsindependently of locally available electrical power.

IP Phonesand VoIP telephone adapters connect toroutersorcable modemswhich typically dependon the availability ofmains electricityor locally generated power.[31]Some VoIP service providersuse customer premises equipment (e.g., cablemodems) with battery-backed power supplies toassure uninterrupted service for up to several hours in case of local power failures. Such battery-backed devices typically are designed for use with analog handsets.

Some VoIP service providers implement services to route calls to other telephone services of thesubscriber, such a cellular phone, in the event that the customer's network device is inaccessible toterminate the call.

The susceptibility of phone service to power failures is a common problem even with traditionalanalog service in areas where many customers purchase modern telephone units that operate withwireless handsets to a base station, or that have other modern phone features, such as built-invoicemail or phone book features.

Redundancy[edit]

The historical separation of IP networks and thePSTNprovided redundancy when no portion of acall was routed over IP network. An IP network outage would not necessarily mean that a voicecommunication outage would occur simultaneously, allowing phone calls to be made during IPnetwork outages. When telephone service relies on IP network infrastructure such as the Internet, anetwork failure can isolate users from all telephony communication, includingEnhanced 911andequivalent services in other locales.[original research?]However, the network design envisioned by

DARPA in the early 1980s included a fault tolerant architecture under adverse conditions.

Security[edit]

The security concerns of VoIP telephone systems are similar to those of any Internet-connecteddevice. This means thathackerswho know about these vulnerabilities can institutedenial-of-serviceattacks, harvest customer data, record conversations and compromise voicemailmessages.[32][33][34]Compromised VoIP user account or session credentials may enable an attacker
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to incur substantial charges from third-party services, such as long-distance or internationaltelephone calling.

The technical details of many VoIP protocols create challenges in routing VoIP trafficthroughfirewallsandnetwork address translators,used to interconnect to transit networks or theInternet. Privatesession border controllersare often employed to enable VoIP calls to and fromprotected networks. Other methods to traverse NAT devices involve assistive protocols suchasSTUNandInteractive Connectivity Establishment(ICE).

Many consumer VoIP solutions do not support encryption of the signaling path or the media,however securing a VoIP phone is conceptually easier to implement than on traditional telephonecircuits. A result of the lack of encryption is a relative easy to eavesdrop on VoIP calls when accessto the data network is possible.[35]Free open-source solutions, such asWireshark,facilitatecapturing VoIP conversations.

Standards for securing VoIP are available in theSecure Real-time Transport Protocol(SRTP) andtheZRTPprotocol foranalog telephony adaptersas well as for somesoftphones.IPsecis availableto securepoint-to-pointVoIP at the transport level by usingopportunistic encryption.

Government and military organizations use various security measures to protect VoIP traffic, such asvoice over secure IP (VoSIP), secure voice over IP (SVoIP), and secure voice over secure IP

(SVoSIP).[36]The distinction lies in whether encryption is applied in the telephone or in thenetwork[37]or both. Secure voice over secure IP is accomplished by encrypting VoIP with protocolssuch asSRTPorZRTP.Secure voice over IP is accomplished by usingType 1 encryptionon aclassified network, likeSIPRNet.[38][39][40][41][42]Public Secure VoIP is also available with free GNUprograms and in many popular commercial VoIP programs via libraries such as ZRTP.[43]

Caller ID[edit]Further information:Caller ID spoofing

Caller IDsupport among VoIP providers varies, but is provided by the majority of VoIP providers.Many VoIP service providers allow callers to configure arbitrary caller ID information, thuspermittingspoofing attacks.[44]Business-grade VoIP equipment and software often makes it easy to

modify caller ID information, providing many businesses great flexibility.The United States enacted theTruth in Caller ID Act of 2009on December 22, 2010. This lawmakes it a crime to "knowingly transmit misleading or inaccurate caller identification information withthe intent to defraud, cause harm, or wrongfully obtain anything of value ...".[45]Rules implementingthe law were adopted by theFederal Communications Commissionon June 20, 2011.[46]

Compatibility with traditional analog telephone sets[edit]

Most analog telephone adapters do not decode dial pulses generated by older telephones,supporting onlytouch-tone.Pulse-to-tone converters are commercially available;[47]a user reportsthat a few specific ATA models (such as theGrandstream 502)recognise pulse dial directly,[48][49]butare poorly documented and provide no assurance that newer models in the same series will retain

this compatibility.

Support for other telephony devices[edit]

Another challenge for VoIP implementations is the proper handling of outgoing calls from othertelephony devices such as digital video recorders, satellite television receivers, alarm systems,conventional modems and other similar devices that depend on access to a PSTN telephone line forsome or all of their functionality.
http://en.wikipedia.org/wiki/Firewall_(networking)http://en.wikipedia.org/wiki/Firewall_(networking)http://en.wikipedia.org/wiki/Firewall_(networking)http://en.wikipedia.org/wiki/Network_address_translatorhttp://en.wikipedia.org/wiki/Network_address_translatorhttp://en.wikipedia.org/wiki/Network_address_translatorhttp://en.wikipedia.org/wiki/Session_border_controllerhttp://en.wikipedia.org/wiki/Session_border_controllerhttp://en.wikipedia.org/wiki/Session_border_controllerhttp://en.wikipedia.org/wiki/STUNhttp://en.wikipedia.org/wiki/STUNhttp://en.wikipedia.org/wiki/STUNhttp://en.wikipedia.org/wiki/Interactive_Connectivity_Establishmenthttp://en.wikipedia.org/wiki/Interactive_Connectivity_Establishmenthttp://en.wikipedia.org/wiki/Interactive_Connectivity_Establishmenthttp://en.wikipedia.org/wiki/Voip#cite_note-35http://en.wikipedia.org/wiki/Voip#cite_note-35http://en.wikipedia.org/wiki/Voip#cite_note-35http://en.wikipedia.org/wiki/Wiresharkhttp://en.wikipedia.org/wiki/Wiresharkhttp://en.wikipedia.org/wiki/Wiresharkhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/Analog_telephony_adapterhttp://en.wikipedia.org/wiki/Analog_telephony_adapterhttp://en.wikipedia.org/wiki/Analog_telephony_adapterhttp://en.wikipedia.org/wiki/Softphonehttp://en.wikipedia.org/wiki/Softphonehttp://en.wikipedia.org/wiki/Softphonehttp://en.wikipedia.org/wiki/IPsechttp://en.wikipedia.org/wiki/IPsechttp://en.wikipedia.org/wiki/IPsechttp://en.wikipedia.org/wiki/Point-to-point_(telecommunications)http://en.wikipedia.org/wiki/Point-to-point_(telecommunications)http://en.wikipedia.org/wiki/Point-to-point_(telecommunications)http://en.wikipedia.org/wiki/Opportunistic_encryptionhttp://en.wikipedia.org/wiki/Opportunistic_encryptionhttp://en.wikipedia.org/wiki/Opportunistic_encryptionhttp://en.wikipedia.org/wiki/Voip#cite_note-36http://en.wikipedia.org/wiki/Voip#cite_note-36http://en.wikipedia.org/wiki/Voip#cite_note-36http://en.wikipedia.org/wiki/Voip#cite_note-37http://en.wikipedia.org/wiki/Voip#cite_note-37http://en.wikipedia.org/wiki/Voip#cite_note-37http://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/Type_1_encryptionhttp://en.wikipedia.org/wiki/Type_1_encryptionhttp://en.wikipedia.org/wiki/Type_1_encryptionhttp://en.wikipedia.org/wiki/SIPRNethttp://en.wikipedia.org/wiki/SIPRNethttp://en.wikipedia.org/wiki/Voip#cite_note-38http://en.wikipedia.org/wiki/Voip#cite_note-38http://en.wikipedia.org/wiki/Voip#cite_note-40http://en.wikipedia.org/wiki/Voip#cite_note-42http://en.wikipedia.org/wiki/Voip#cite_note-42http://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/Voip#cite_note-43http://en.wikipedia.org/wiki/Voip#cite_note-43http://en.wikipedia.org/wiki/Voip#cite_note-43http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=16http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=16http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=16http://en.wikipedia.org/wiki/Caller_ID_spoofinghttp://en.wikipedia.org/wiki/Caller_ID_spoofinghttp://en.wikipedia.org/wiki/Caller_ID_spoofinghttp://en.wikipedia.org/wiki/Caller_IDhttp://en.wikipedia.org/wiki/Caller_IDhttp://en.wikipedia.org/wiki/Spoofing_attackhttp://en.wikipedia.org/wiki/Spoofing_attackhttp://en.wikipedia.org/wiki/Voip#cite_note-44http://en.wikipedia.org/wiki/Voip#cite_note-44http://en.wikipedia.org/wiki/Voip#cite_note-44http://en.wikipedia.org/w/index.php?title=Truth_in_Caller_ID_Act_of_2009&action=edit&redlink=1http://en.wikipedia.org/w/index.php?title=Truth_in_Caller_ID_Act_of_2009&action=edit&redlink=1http://en.wikipedia.org/w/index.php?title=Truth_in_Caller_ID_Act_of_2009&action=edit&redlink=1http://en.wikipedia.org/wiki/Voip#cite_note-45http://en.wikipedia.org/wiki/Voip#cite_note-45http://en.wikipedia.org/wiki/Voip#cite_note-45http://en.wikipedia.org/wiki/Federal_Communications_Commissionhttp://en.wikipedia.org/wiki/Federal_Communications_Commissionhttp://en.wikipedia.org/wiki/Federal_Communications_Commissionhttp://en.wikipedia.org/wiki/Voip#cite_note-46http://en.wikipedia.org/wiki/Voip#cite_note-46http://en.wikipedia.org/wiki/Voip#cite_note-46http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=17http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=17http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=17http://en.wikipedia.org/wiki/Touch-tonehttp://en.wikipedia.org/wiki/Touch-tonehttp://en.wikipedia.org/wiki/Touch-tonehttp://en.wikipedia.org/wiki/Voip#cite_note-47http://en.wikipedia.org/wiki/Voip#cite_note-47http://en.wikipedia.org/wiki/Voip#cite_note-47http://en.wikipedia.org/wiki/Grandstream_Networkshttp://en.wikipedia.org/wiki/Grandstream_Networkshttp://en.wikipedia.org/wiki/Grandstream_Networkshttp://en.wikipedia.org/wiki/Voip#cite_note-48http://en.wikipedia.org/wiki/Voip#cite_note-48http://en.wikipedia.org/wiki/Voip#cite_note-48http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=18http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=18http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=18http://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=18http://en.wikipedia.org/wiki/Voip#cite_note-48http://en.wikipedia.org/wiki/Voip#cite_note-48http://en.wikipedia.org/wiki/Grandstream_Networkshttp://en.wikipedia.org/wiki/Voip#cite_note-47http://en.wikipedia.org/wiki/Touch-tonehttp://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=17http://en.wikipedia.org/wiki/Voip#cite_note-46http://en.wikipedia.org/wiki/Federal_Communications_Commissionhttp://en.wikipedia.org/wiki/Voip#cite_note-45http://en.wikipedia.org/w/index.php?title=Truth_in_Caller_ID_Act_of_2009&action=edit&redlink=1http://en.wikipedia.org/wiki/Voip#cite_note-44http://en.wikipedia.org/wiki/Spoofing_attackhttp://en.wikipedia.org/wiki/Caller_IDhttp://en.wikipedia.org/wiki/Caller_ID_spoofinghttp://en.wikipedia.org/w/index.php?title=Voice_over_IP&action=edit&section=16http://en.wikipedia.org/wiki/Voip#cite_note-43http://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/Voip#cite_note-42http://en.wikipedia.org/wiki/Voip#cite_note-40http://en.wikipedia.org/wiki/Voip#cite_note-40http://en.wikipedia.org/wiki/Voip#cite_note-38http://en.wikipedia.org/wiki/Voip#cite_note-38http://en.wikipedia.org/wiki/SIPRNethttp://en.wikipedia.org/wiki/Type_1_encryptionhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Voip#cite_note-37http://en.wikipedia.org/wiki/Voip#cite_note-36http://en.wikipedia.org/wiki/Opportunistic_encryptionhttp://en.wikipedia.org/wiki/Point-to-point_(telecommunications)http://en.wikipedia.org/wiki/IPsechttp://en.wikipedia.org/wiki/Softphonehttp://en.wikipedia.org/wiki/Analog_telephony_adapterhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Wiresharkhttp://en.wikipedia.org/wiki/Voip#cite_note-35http://en.wikipedia.org/wiki/Interactive_Connectivity_Establishmenthttp://en.wikipedia.org/wiki/STUNhttp://en.wikipedia.org/wiki/Session_border_controllerhttp://en.wikipedia.org/wiki/Network_address_translatorhttp://en.wikipedia.org/wiki/Firewall_(networking)


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These types of calls sometimes complete without any problems, but in other cases they fail. If VoIPandcellularsubstitution becomes very popular, some ancillary equipment makers may be forced toredesign equipment, because it would no longer be possible to assume a conventional PSTNtelephone line would be available in consumers' houses.

User and administrative interfaces[edit]

Voice over IP services typically take advantage of other Internet- or web-based facilities foroperation and administration. Websites provide customer interaction, account configuration, servicestatistic

voice over ip

Documents

ip telephony

ip communications

internet telephony

legacy networks

early providers of voice

ip packets

closed networks

communication networks