1974  The Institute of Electrical and Electronic Engineers (IEEE) published a paper titled “A Protocol for Packet Network Interconnection.”

1981  IPv4 is described in RFC 791.

1985  The National Science Foundation commissions the creation of NSFNET.

1995  VocalTec releases the first commercial Internet phone software.

1996 

ITU-T begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the H.323 standard.

US telecommunication companies petition the US Congress to ban Internet phone technology.

1997  Level 3 began development of its first softswitch, a term they coined in 1998.

1999 

The Session Initiation Protocol (SIP) specification RFC 2543 is released.

Mark Spencer of Digium develops the first open source Private branch exchange (PBX) software (Asterisk).

2004  Commercial VoIP service providers proliferate.

2005  OpenSER (later Kamailio and OpenSIPS) SIP proxy server is forked from the SIP Express Router.

2006  FreeSWITCH open source software is released.

VoIP technologies and implementations

Voice-over-IP has been implemented in various ways using both proprietary and open protocols and standards. Examples of technologies used to implement Voice over Internet Protocol include:

H.323

IP Multimedia Subsystem (IMS)

Media Gateway Control Protocol (MGCP)

Session Initiation Protocol (SIP)

Real-time Transport Protocol (RTP)

The Session Initiation Protocol has gained wide-spread VoIP market penetration, while H.323 deployments are increasingly limited to carrying existing long-haul network traffic.[citation needed]

A notable proprietary implementation is the Skype network. Other examples of specific implementations and a comparison between them are available in Comparison of VoIP software.

Adoption

Consumer market

Example of VoIP adapter setup in residential network

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 domestic calling, and some to other countries as well, for a flat monthly fee as well as free calling between subscribers using the same provider. These services have a wide variety of features which can be more or less similar to traditional POTS.

There are three common methods of connecting to VoIP service providers:

A typical analog telephone adapter (ATA) for connecting an analog phone to a VoIP provider

An Analog Telephone Adapter (ATA) may be connected between an IP network (such as a broadband connection) and an existing telephone jack in order to provide service nearly indistinguishable from PSTN providers on all the other telephone 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.

Dedicated VoIP phones are phones that allow VoIP calls without the use of a computer. Instead they connect directly to the IP network (using technologies such as Wi-Fi or Ethernet). In order to connect to the PSTN they usually require service from a VoIP service provider; most people therefore will use them in conjunction with a paid service plan.

A softphone (also known as an Internet phone or Digital phone) is a piece of software that can be installed on a computer that allows VoIP calling without dedicated hardware.

PSTN and mobile network providers

It is becoming increasingly common for telecommunications providers to use VoIP telephony over dedicated and public IP networks to connect switching stations and to interconnect with other telephony network providers; this is often referred to as “IP backhaul”.

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 phone networks.

“Dual mode” telephone sets, which allow for the seamless handover between a cellular network and a Wi-Fi network, are expected to help VoIP become more popular.

Phones such as the NEC N900iL, many of the Nokia Eseries and several other Wi-Fi enabled mobile phones have SIP clients built into the firmware. Such clients operate independently of the mobile phone network (however some operators choose to remove the client from subsidised handsets). Some operators such as Vodafone actively try to block VoIP traffic from their network. Others, like T-Mobile, have refused to interconnect with VoIP-enabled networks as was seen in the legal case between T-Mobile and Truphone, which ultimately was settled in the UK High Court in favour of the VoIP carrier.

Corporate use

Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses are gradually beginning to migrate from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs.

VoIP solutions aimed at businesses have evolved into “unified communications” services that treat all communicationshone calls, faxes, voice mail, e-mail, Web conferences and mores discrete units that can all be delivered via any means and to any handset, including cellphones. Two kinds of competitors 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.

VoIP runs both voice and data communications over a single network, which can significantly reduce infrastructure costs.

The prices of extensions on VoIP are lower than for PBXs and key systems. VoIP switches run on commodity hardware, such as PCs or Linux systems. Rather than closed architectures, these devices rely on standard interfaces.

VoIP devices have simple, intuitive user interfaces, so users can often make simple system configuration changes. Dual-mode cellphones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Fi network, so that it is no longer necessary to carry both a desktop phone and a cellphone. Maintenance becomes simpler as there are fewer devices to oversee.

Skype, which originally marketed itself as a service among friends, has begun to cater to businesses, providing free-of-charge connection between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge.

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

Benefits

Operational cost

VoIP can be a benefit for reducing communication and infrastructure costs. Examples include:

Routing phone calls over existing data networks to avoid the need for separate voice and data networks.

Conference calling, IVR, call forwarding, automatic redial, and caller ID features that traditional telecommunication companies (telcos) normally charge extra for are available free of charge from open source VoIP implementations.

Costs are lower, mainly because of the way Internet access is billed compared to regular telephone calls. While regular telephone calls are billed by the minute or second, VoIP calls are billed per megabyte (MB). In other words, VoIP calls are billed per amount of information (data) sent over the Internet and not according to the time connected to the telephone network. In practice the amount charged for the data transferred in a given period is far less than that charged for the amount of time connected on a regular telephone line.

Flexibility

VoIP can facilitate tasks and provide services that may be more difficult to implement using the PSTN. Examples include:

The ability to transmit more than one telephone call over a single broadband connection without the need to add extra lines.

Secure calls using standardized protocols (such as Secure Real-time Transport Protocol). Most of the difficulties of creating a secure telephone connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream.

Location independence. Only a sufficiently fast and stable Internet connection is needed to get a connection from anywhere to a VoIP provider.

Integration with other services available over the Internet, including video conversation, message or data file exchange during the conversation, audio conferencing, managing address books, and passing information about whether other people are available to interested parties.

Challenges

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Quality of service

By default, IP routers handle traffic on a first-come, first-served basis. When a packet is routed to a link where another packet is already being sent, the router holds it on a queue. Should additional traffic arrive faster than the queued traffic can be sent, the queue will grow. If VoIP packets have to wait their turn in a long queue, intolerable latency may result.

One way to avoid this problem is to simply ensure that the links are fast enough so that queues never build even in the worst case. This usually requires additional mechanisms to limit the amount of traffic entering the network, and for voice traffic this is usually done by limiting the number of simultaneous calls. Another approach is to use quality-of-service (QoS) mechanisms such as Diffserv to give priority to VoIP packets and other latency-sensitive traffic so they can “jump the line” and be transmitted ahead of any bulk data packets already in the queue. This can work quite well when voice constitutes a relatively small fraction of the total network load, as it usually does in today’s Internet.

Generally a VoIP packet still has to wait for the current packet to finish transmission; although it is possible to pre-empt (abort) a less important packet in mid-transmission, this is not commonly done, especially on high speed links where transmission times are small even for maximum-sized packets. An alternative to pre-emption on slower links, such as dialup and DSL, is to reduce the maximum transmission time by reducing the maximum transmission unit. But every packet must contain protocol headers, so this increases relative header overhead on every link along that user’s Internet paths, not just the bottleneck link (which is usually his Internet access link.)

ADSL modems invariably provide Ethernet (or Ethernet over USB) connections to local equipment, but inside they are actually ATM modems. They use AAL5 to segment each Ethernet packet into a series of 48-byte ATM cells for transmission and reassemble them back into Ethernet packets at the receiver. A virtual circuit identifier (VCI) is part of the 5-byte header on every ATM cell, so the transmitter can multiplex the active VCs in any arbitrary order. (Cells from the same VC are always sent sequentially.)

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

It should be noted that this doesn’t come for free. ATM has substantial header overhead: 5/53 = 9.4%, roughly twice the total header overhead of a 1500 byte TCP/IP/Ethernet packet (with TCP timestamps). This “ATM tax” is incurred by every DSL user whether or not he takes advantage of multiple virtual circuits – and few can.

ATM’s potential for latency reduction is greatest on slow links, because worst-case latency decreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kb/s but only 8 ms at 1.5 Mb/s. If this is the bottleneck link, this latency is probably small enough to ensure good VoIP performance without MTU reductions or multiple ATM PVCs. The latest generations of DSL, VDSL and VDSL2, carry Ethernet without intermediate ATM/AAL5 layers, and they generally support IEEE 802.1p priority tagging so that VoIP can be queued ahead of less time-critical traffic.

Voice, and all other data, travel in packets over IP networks with fixed maximum capacity. This system is more prone to congestion[citation needed] and DoS attacks than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.

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 to a geostationary satellite and back; delays of 400-600 ms are typical.

When the load on a link grows so quickly that its queue overflows, congestion results and data packets are lost. This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion. But VoIP usually does not use TCP because recovering from congestion through retransmission usually entails too much latency. So QoS mechanisms can avoid the undesirable loss of VoIP packets by immediately transmitting them ahead of any queued bulk traffic on the same link, even when that bulk traffic queue is overflowing.

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. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing analog audio, although this further increases delay. This avoids a condition known as buffer underrun, in which the voice engine is missing audio since the next voice packet has not yet arrived. When IP packets are lost or delayed at any point in the network between VoIP users there will be a momentary dropout of voice if all packet delay and loss mechanisms cannot compensate.

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, 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.[citation needed]

A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP Extended Report (RFC 3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC 3611 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 (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 related to the jitter buffer.

RFC 3611 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. RFC 3611 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.

Layer-2 quality of service

A number of protocols that deal with the data link layer and physical layer include quality-of-service mechanisms that can be used to ensure that applications like VoIP work well even in congested scenarios. Some examples include:

IEEE 802.11e is an approved amendment to the IEEE 802.11 standard that defines a set of quality-of-service enhancements for wireless LAN applications through modifications to the Media Access Control (MAC) layer. The standard is considered of critical importance for delay-sensitive applications, such as Voice over Wireless IP.

IEEE 802.1p defines 8 different classes of service (including one dedicated to voice) for traffic on layer-2 wired Ethernet.

The ITU-T G.hn standard, which provides a way to create a high-speed (up to 1 gigabit per second) Local area network using existing home wiring (power lines, phone lines and coaxial cables). G.hn provides QoS by means of “Contention-Free Transmission Opportunities” (CFTXOPs) which are allocated to flows (such as a VoIP call) which require QoS and which have negotiated a “contract” with the network controller.

Susceptibility to power failure

Telephones for traditional residential analog service are usually connected directly to telephone company phone lines which provide direct current to power most basic analog handsets independently of locally available power.

IP Phones and VoIP telephone adapters connect to routers or cable modems which typically depend on the availability of mains electricity or locally generated power. Some VoIP service providers use customer premise equipment (e.g., cablemodems) with battery-backed power supplies to assure 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.

The susceptibility of phone service to power failures is a common problem even with traditional analog service in areas where many customers purchase modern handset units that operate wirelessly to a base station, or that have other modern phone features, such as built-in voicemail or phone book features.

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. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department. In the United States, at least one major police department has strongly objected to this practice as potentially endangering the public.

A fixed line phone has a direct relationship between a telephone number and a physical location. A telephone number represents one pair of wires that links a location to the telephone company’s exchange. Once a line is connected, the telephone company stores the home address that relates to the wires, and this relationship will rarely change. If an emergency call comes from that number, then the physical location is known.

In the IP world, it is not so simple. A broadband provider may know the location where the wires terminate, but this does not necessarily allow the mapping of an IP address to that location. IP addresses are often dynamically assigned, so the ISP may allocate an address for online access, or at the time a broadband router is engaged. The ISP recognizes individual IP addresses, but does not necessarily know what physical location to which it corresponds. The broadband service provider knows the physical location, but is not necessarily tracking the IP addresses in use.

There are more complications, since IP allows a great deal of mobility. For example, a broadband connection can be used to dial a virtual private network that is employer-owned. When this is done, the IP address being used will belong to the range of the employer, rather than the address of the ISP, so this could be many kilometres away or even in another country. To provide another example: if mobile data is used, e.g., a 3G mobile handset or USB wireless broadband adapter, then the IP address has no relationship with any physical location, since a mobile user could be anywhere that there is network coverage, even roaming via another cellular company.

In short, there is no relationship between IP address and physical location, so the address itself reveals no useful information for the emergency services.

At the VoIP level, a phone or gateway may identify itself with a SIP registrar by using a username and password. So in this case, the Internet Telephony Service Provider (ITSP) knows that a particular user is online, and can relate a specific telephone number to the user. However, it does not recognize how that IP traffic was engaged. Since the IP address itself does not necessarily provide location information presently, today a “best efforts” approach is to use an available database to find that user and the physical address the user chose to associate with that telephone numberlearly an imperfect solution.

VoIP Enhanced 911 (E911) is another method by which VoIP providers in the United States are able to support emergency services. The VoIP E911 emergency-calling system associates a physical address with the calling party’s telephone number as required by the Wireless Communications and Public Safety Act of 1999. All “interconnected” VoIP providers (those that provide access to the PSTN system) are required to have E911 available to their customers. VoIP E911 service generally adds an additional monthly fee to the subscriber’s service per line, similar to analog phone service. Participation in E911 is not required and customers can opt-out or disable E911 service on their VoIP lines, if desired. VoIP E911 has been successfully used by many VoIP providers to provide physical address information to emergency service operators.

One shortcoming of VoIP E911 is that the emergency system is based on a static table lookup. Unlike in cellular phones, where the location of an E911 call can be traced using Assisted GPS or other methods, the VoIP E911 information is only accurate so long as subscribers are diligent in keeping their emergency address information up-to-date. In the United States, the Wireless Communications and Public Safety Act of 1999 leaves the burden of responsibility upon the subscribers and not the service providers to keep their emergency information up to date.

Lack of redundancy

With the current separation of the Internet and the PSTN, a certain amount of redundancy is provided. An Internet outage does not necessarily mean that a voice communication outage will occur simultaneously, allowing individuals to call for emergency services and many businesses to continue to operate normally. In situations where telephone services become completely reliant on the Internet infrastructure, a single-point failure can isolate communities from all communication, including Enhanced 911 and equivalent services in other locales.

Number portability

Local number portability (LNP) and Mobile number portability (MNP) also impact VoIP business. In November 2007, the Federal Communications Commission in the United States released an order extending number portability obligations to interconnected VoIP providers and carriers that support VoIP providers. Number portability is a service that allows a subscriber to select a new telephone carrier without requiring a new number to be issued. Typically, it is the responsibility of the former carrier to “map” the old number to the undisclosed number assigned by the new carrier. This is achieved by maintaining a database of numbers. A dialed number is initially received by the original carrier and quickly rerouted to the new carrier. Multiple porting references must be maintained even if the subscriber returns to the original carrier. The FCC mandates carrier compliance with these consumer-protection stipulations.

A voice call originating in the VoIP environment also faces challenges to reach its destination if the number is routed to a mobile phone number on a traditional mobile carrier. VoIP has been identified in the past as a Least Cost Routing (LCR) system, which is based on checking the destination of each telephone call as it is made, and then sending the call via the network that will cost the customer the least. This rating is subject to some debate given the complexity of call routing created by number portability. With GSM number portability now in place, LCR providers can no longer rely on using the network root prefix to determine how to route a call. Instead, they must now determine the 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 a central database, like the UK, it might be necessary to query the GSM network about which home network a mobile phone number belongs to. As the popularity of VoIP increases in the enterprise markets because of least cost routing options, it needs to provide a certain level of reliability when handling calls.

MNP checks are important to assure that this quality of service is met. By handling MNP lookups before routing a call and by assuring that the voice call will actually work, VoIP service providers are able to offer business subscribers the level of reliability they require.

PSTN integration

E.164 is a global numbering standard for both the PSTN and PLMN. Most VoIP implementations support E.164 to allow calls to be routed to and from VoIP subscribers and the PSTN/PLMN. VoIP implementations can also allow other identification techniques to be used. For example, Skype allows subscribers to choose “Skype names” (usernames) whereas SIP implementations can use URIs similar to email addresses. Often VoIP implementations employ methods of translating non-E.164 identifiers to E.164 numbers and vice-versa, such as the Skype-In service provided by Skype and the ENUM service in IMS and SIP.

Echo can also be an issue for PSTN integration . Common causes of echo include impedance mismatches in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end.

Security

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Voice over Internet Protocol telephone systems (VoIP) are susceptible to attacks as are any internet-connected devices. This means that hackers who know about these vulnerabilities can institute denial-of-service attacks, harvest customer data, record conversations and break into voice mailboxes.

Another challenge is routing VoIP traffic through firewalls and network address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from protected networks. 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 NATs involve using protocols such as STUN or ICE.

Many consumer VoIP solutions do not support encryption, 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. An attacker with a packet sniffer could intercept your VoIP calls if you are not on a secure VLAN.

There are open source solutions, such as Wireshark, that facilitate sniffing of VoIP conversations. A modicum of security is afforded by patented audio codecs in proprietary implementations that are not easily available for open source applications[citation needed], however such security through obscurity has not proven effective in other fields.[citation needed] Some vendors also use compression to make eavesdropping more difficult.[citation needed] However, real security requires encryption and cryptographic authentication which are not widely supported at a consumer level. The existing security standard Secure Real-time Transport Protocol (SRTP) and the new ZRTP protocol are 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[citation needed]. In 2005, Skype invited a researcher, Dr Tom Berson, to assess the security of the Skype software, and his conclusions are available in a published report.

The Voice VPN solution provides secure voice for enterprise VoIP networks by applying IPSec encryption to the digitized voice stream.

Securing VoIP

To prevent the above security concerns the government and military organizations are using; Voice over Secure IP (VoSIP), Secure Voice over IP (SVoIP), and Secure Voice over Secure IP (SVoSIP) to protect confidential, and/or classified VoIP communications. Secure Voice over IP is accomplished by encrypting VoIP with Type 1 encryption. Secure Voice over Secure IP is accomplished by using Type 1 encryption on a classified network, like SIPRNet. Public Secure VoIP is also available with free GNU programs.

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.

In a few cases, VoIP providers may allow a caller to spoof the Caller ID information, potentially making calls appear as though they are from a number that does not belong to the caller 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.

The “Truth in Caller ID Act” has been in preparation in the US Congress since 2006, but as of January 2009 still has not been enacted. This bill proposes to make it a crime in the United States to “knowingly transmit misleading or inaccurate caller identification information with the intent to defraud, cause harm, or wrongfully obtain anything of value …”

Compatibility with traditional analog telephone sets

Some analog telephone adapters do not decode pulse dialing from older phones. They may only work with push-button telephones using the touch-tone system. The VoIP user may use a pulse-to-tone converter, if needed.

Fax handling

Support for sending faxes over VoIP implementations is still limited. The existing voice codecs are not designed for fax transmission; they are designed to digitize an analog representation of a human voice efficiently. However, the inefficiency of digitizing an analog representation (modem signal) of a digital representation (a document image) of analog data (an original document) more than negates any bandwidth advantage of VoIP. In other words, the fax “sounds” simply don’t fit in the VoIP channel. An alternative IP-based solution for delivering fax-over-IP called T.38 is available.

The T.38 protocol is designed compensate for the differences between traditional packet-less communications over analog lines and packet based transmissions which are the basis for IP communications. 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. Originally, T.38 was designed to use UDP and TCP transmission methods across an IP network. TCP is better suited for use between two IP devices. However, older fax machines, connected to an analog system, benefit from UDP near real-time characteristics due to the “no recovery rule” when a UDP packet is lost or an error occurs during transmission. UDP transmissions are preferred as they do not require testing for dropped packets and as such since each T.38 packet transmission includes a majority of the data sent in the prior packet, a T.38 termination point has a higher degree of success in re-assembling the fax transmission back into its original form for interpretation by the end device. This in an attempt to overcome the obstacles of simulating real time transmissions using packet based protocol.

There have been updated versions of T.30 to resolve the fax over IP issues, which is the core fax protocol. Some newer high end fax machines have T.38 built-in capabilities which allow the user to plug right into the network and transmit/receive faxes in native T.38 like the Ricoh 4410NF Fax Machine. A unique feature of T.38 is that each packet contains a portion of the main data sent in the previous packet. With T.38, two successive lost packets are needed to actually lose any data. The data you lose will only be a small piece, but with the right settings and error correction mode, there is an increased likelihood that you will receive enough of the transmission to satisfy the requirements of the fax machine for output of the sent document.

Support for other telephony devices

Another challenge for VoIP implementations is the proper handling of outgoing calls from other telephony devices such as DVR boxes, satellite television receivers, alarm systems, conventional modems and other similar devices that depend on access to a PSTN telephone line for some or all of their functionality.

These types of calls sometimes complete without any problems, but in other cases they fail. 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 PSTN telephone line would be available in consumer’s homes.

Legal issues

As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers, governments are becoming more interested in regulating VoIP in a manner similar to PSTN services.

Another legal issue that the US Congress is debating concerns changes to the Foreign Intelligence Surveillance Act. The issue in question is calls between Americans and foreigners. The National Security Agency (NSA) isn’t authorized to tap Americans’ conversations without a warrantut the Internet, and specifically VoIP doesn’t draw as clear a line to the location of a caller or a call’s recipient as the traditional phone system does. As VoIP’s low cost and flexibility convinces more and more organizations to adopt the technology, the line separating the NSA’s ability to snoop on phone calls will only get blurrier. VoIP technology has also increased security concerns because VoIP and similar technologies have made it more difficult for the government to determine where a target is physically located when communications are being intercepted, and that creates a whole set of new legal challenges.

In the US, the Federal Communications Commission now requires all interconnected VoIP service providers to comply with requirements comparable to those for traditional telecommunications service providers. VoIP operators in the US are required to support local number portability; make service accessible to people with disabilities; pay regulatory fees, universal service contributions, and other mandated payments; and enable law enforcement authorities to conduct surveillance pursuant to the Communications Assistance for Law Enforcement Act (CALEA). “Interconnected” VoIP operators also must provide Enhanced 911 service, disclose any limitations on their E-911 functionality to their consumers, and obtain affirmative acknowledgements of these disclosures from all consumers. VoIP operators also receive the benefit of certain US telecommunications regulations, including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. Providers of “nomadic” VoIP service  those who are unable to determine the location of their users  are exempt from state telecommunications regulation.

Throughout the developing world, countries where regulation is weak or captured by the dominant operator, restrictions on the use of VoIP are imposed, including in Panama where VoIP is taxed, Guyana where VoIP is prohibited and India where its retail commercial sales is allowed but only for long distance service. 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.

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.

In the UAE, it is illegal to use any form of VoIP, to the extent that Web sites of Skype and Gizmo5 are blocked.

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 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 US-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 US VoIP providers.

International VoIP implementation

IP telephony in Japan

This section does not cite any references or sources.

Please help improve this article by adding citations to reliable sources. Unsourced material may be challenged and removed. (January 2009)

In Japan, IP telephony (IP, IP Denwa ?) is regarded as a service applied by VoIP technology to the whole or a part of the telephone line. As of 2003, IP telephony services have been assigned telephone numbers. IP telephony services also often include videophone/video conferencing services. 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 contracts with customers, and have an obligation to respond to their complaints cordially.

Many Japanese Internet service providers (ISP) are including IP telephony services. An ISP who also provides IP telephony service is known as a “ITSP (Internet Telephony Service Provider)”. Recently, the competition among ITSPs has been activated, by option or set sales, in connection with ADSL or FTTH services.

The tariff system normally applied to Japanese IP telephony is described below;

A call between IP telephony subscribers, limited to the same group, is usually free of charge.

A call from IP telephony subscribers to a fixed line or PHS is usually a uniformly fixed rate all over the country.

Between ITSPs, the interconnection is mostly maintained at VoIP level.

Where the IP telephony is assigned normal telephone number (0AB-J), the condition for its interconnection is considered same as normal telephony.

Where the IP telephony is assigned specific telephone number (050), the condition for its interconnection is described below;

Interconnection is sometimes charged. (Sometimes, it is free of charge.) In case of free-of-charge, mostly, communication traffic is exchanged via a P2P connection with the same VoIP standard. Otherwise, certain conversions are needed at the point of the VoIP gateway which incurs operating costs.

Since September 2002, the MIC has assigned IP telephony telephone numbers on the condition that the service falls into certain required categories of quality.

High-quality IP telephony is assigned a telephone number, normally starting with the digits 050. When VoIP quality is so high that a customer has difficulty telling 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.

See also

Wikimedia Commons has media related to: VoIP

Audio over IP

Capillary routing

Click to talk

Communications Assistance For Law Enforcement Act

Comparison of VoIP software

Computer conferencing

Differentiated services

ENUM

H.323

Harvard sentences

Integrated services

Internet fax

Push-button telephone

IP Multimedia Subsystem

IP Phone

Mobile VoIP

Mouth-to-ear delay

Network Voice Protocol

Predictive dialers

Publicly Available Telephone Services

RTP audio video profile

Secure telephone

Session Initiation Protocol (SIP)

TestYourVoIP

Voice VPN

VoiceXML

VoIP Dial Plan

VoIP recording

References

^ “Voice over Internet Protocol. Definition and Overview”. International Engineering Consortium. 2007. http://www.iec.org/online/tutorials/int_tele/index.asp. Retrieved 2009-04-27. 

^ Vinton G. Cerf, Robert E. Kahn, “A Protocol for Packet Network Intercommunication”, IEEE Transactions on Communications, Vol. 22, No. 5, May 1974 pp. 637-648

^ “The Launch of NSFNET”. The National Science Foundation. http://www.nsf.gov/about/history/nsf0050/internet/launch.htm. Retrieved 2009-01-21. 

^ Keating, Tom. “Internet Phone Release 4″ (PDF). Computer Telephony Interaction Magazine. http://blog.tmcnet.com/blog/tom-keating/docs/cti-buyers-guide-1996.pdf. Retrieved 2007-11-07. 

^ “The 10 that Established VoIP (Part 1: VocalTec)”. iLocus. http://www.ilocus.com/2007/07/the_10_that_established_voip_p_2.html. Retrieved 2009-01-21. 

^ “H.323 Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service”. ITU-T. http://www.itu.int/rec/T-REC-H.323-199611-S/en. Retrieved 2009-01-21. 

^ “RFC 2235″. R. Zakon. http://www.faqs.org/rfcs/rfc2235.html. Retrieved 2009-01-21. 

^ “The 10 that Established VoIP (Part 2: Level 3)”. iLocus. July 13, 2007. http://www.ilocus.com/2007/07/the_10_that_established_voip_p_1.html. Retrieved 2007-11-07. 

^ “RFC 2543, SIP: Session Initiation Protocol”. Handley,Schulzrinne,Schooler,Rosenberg. http://www.ietf.org/rfc/rfc2543.txt. Retrieved 2009-01-21. 

^ “What is Asterisk”. Asterisk.org. http://www.asterisk.org/about. Retrieved 2009-01-21. 

^ a b “VoIP on the Verge”. Telecommuncations Online. November 1, 2004. http://www.telecoms-mag.com/Americas/article.asp?HH_ID=AR_539. Retrieved 2009-01-21. 

^ “VoIP: An end to international tariffs?”. Telecommunications Online. http://www.telecoms-mag.com/Americas/article.asp?HH_ID=AR_763. Retrieved 2009-01-21. 

^ “Carriers look to IP for backhaul”. Telecommunications Online. January 21, 2009. http://www.commsdesign.com/showArticle.jhtml;jsessionid=YBQXDJILLQX0IQSNDLPSKH0CJUNN2JVN?articleID=159907138. Retrieved 2009-01-21. 

^ “Mobile’s IP challenge”. Total Telecom. December 8, 2005. http://www.totaltele.com/View.aspx?ID=77588&t=4. Retrieved 2009-01-21. 

^ “Dual-mode cellular/WiFi handset adoption”. TMCnet. http://blog.tmcnet.com/blog/tom-keating/voip/dualmode-cellularwifi-handset-adoption.asp. Retrieved 2006-05-26. 

^ Vodafone Terms and Conditions & Mobile Phones from Phones 4u

^ NEWS.BBC.co.uk, “T-Mobile must open Truphone lines”  BBC

^ “VoIP Review”. VoIP Review, LLC. http://www.voipreview.org. Retrieved 2008-12-06. 

^ Callahan, Renee (December 9, 2008). “Businesses Move To Voice-Over-IP”. forbes.com. http://www.forbes.com/2008/12/09/skype-vonage-ringcentral_leadership_clayton_in_rc_1209claytonchristensen_inl.html. Retrieved 2009-03-03. 

^ a b c Korzeniowski, Peter (January 8, 2009). “Three Technologies You Need In 2009″. Forbes.com. http://www.forbes.com/2009/01/08/small-business-voip-ent-tech-cx_bm_0108bmightytech09.html/. Retrieved 2009-03-02. 

^ “Skype For Business”. skype.com. http://www.skype.com/business/allfeatures/3skypephone/. Retrieved 2009-03-16. 

^ William Jackson (2009-05-27). “SSA goes big on VOIP”. Government Computer News. http://gcn.com/Articles/2009/06/01/SSA-VOIP-implementation.aspx?s=gcndaily_280509&Page=1. Retrieved 2009-05-28. 

^ “Social Security to Build “World’s Largest VoIP”". Government Technology. http://www.govtech.com/gt/275677. Retrieved 2009-05-29. 

^ FCC.gov, What are some advantages of VoIP?

^ FCC.gov, If I have VoIP service, who can I call?

^ Creating a secure and reliable VoIP solution (section: Protecting VoIP at the application layer)  ZDnet Asia

^ VoIP  Vulnerability over Internet Protocol

^ IEEE Multipath routing with adaptive playback scheduling for Voice over IP in Service Overlay Networks (Abstract)

^ ICT Regulation Tool Kit  4.4 VoIP  Regulatory Issues  Universal Service

^ Letter from the City of New York to the Federal Communications Commission

^ FCC Consumer Advisory: “VoIP and 911 Service”

^ It is a common misconception that the Internet was designed for data communications redundancy in the event of war or large-scale natural disaster. In fact, its predecessor, Arpanet, was designed for cost reduction in order to allow universities to share computing resources underwritten by the US Department of Defense.

^ Keeping your telephone number when you change your service provider  FCC

^ VoIpservice.com

^ “RFC 3824  Using E.164 numbers with the Session Initiation Protocol (SIP)”. The Internet Society. June 1, 2004. http://www.packetizer.com/rfc/rfc3824/. Retrieved 2009-01-21. 

^ “Create a Skype Name”. Skype. http://www.skype.com/help/guides/createskypename_windows/. Retrieved 2009-01-21. 

^ “RFC 3969  The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP)”. The Internet Society. December 1, 2004. http://www.packetizer.com/rfc/rfc3969/. Retrieved 2009-01-21. 

^ “Your personal online number”. Skype. http://www.skype.com/allfeatures/onlinenumber/. Retrieved 2009-01-21. 

^ “Application-level Network Interoperability and the Evolution of IMS”. TMCnet.com. May 24, 2006. http://ipcommunications.tmcnet.com/hot-topics/MCP/articles/1311-application-level-network-interoperability-the-evolution-ims.htm. Retrieved 2009-01-21. 

^ Packetcable Implementation P557  Jeff Riddel  ISBN 1587051818 Google Books Preview

^ Taub, Eric (April 2, 2008). “VoIP System Security: Time to Worry, or Maybe Not”. New York Times. http://bits.blogs.nytimes.com/2008/04/02/voip-system-security-time-to-worry-or-maybe-not/?scp=4&sq=voip&st=cse/. Retrieved 2009-03-02. 

^ “Examining Two Well-Known Attacks on VoIP”. CircleID. http://www.circleid.com/posts/examining_two_well_known_attacks_on_voip1/. Retrieved 2006-04-05. 

^ Skyp.com, “Skype Security Evaluation”, Tom Berson/Anagram Laboratories

^ Disa.mil, Internet Protocol Telephony & Voice over Internet Protocol Security Technical Implementation Guide

^ IJCSNS.org, Secure Voice-over-IP

^ Sans.org, SANS Institute InfoSec Reading Room

^ JHU.edu, Packet Loss Concealment in a Secure Voice over IP Environment

^ GDC4S.com, State-of-the-art voice over IP encryptor

^ Networkworld.com, Cellcrypt secure VoIP heading to BlackBerry.

^ Freesorftwaremagazine.com, Secure VoIP calling, free software, and the right to privacy

^ VoIPSA.org, Blog: “Hello Mom, I’m a Fake!” (Telespoof and Fakecaller).

^ Govtrack.us, “Truth in Caller ID Act”.

^ Oldphoneworks.com

^ Soft-Switch.org, Faxing over IP networks

^ Umass.edu, UMass Discussion on UDP transmission Characteristics.

^ Faqs.org, RFC 3362-T.38

^ FaxSuperstore.com, Ricoh 4410NF

^ “Global VoIP Policy Status Matrix”. Global IP Alliance. http://www.ipall.org/matrix/. Retrieved 2006-11-23. 

^ Greenberg, Andy (May 15, 2008). “The State Of Cybersecurity Wiretapping’s Fuzzy Future”. www.forbes.com. http://www.forbes.com/2008/05/15/wiretapping-voip-lichtblau-tech-security08-cx_ag_0515wiretap.html/. Retrieved 2009-03-02. 

^ Greenberg, Andy (May 15, 2008). “The State Of Cybersecurity Wiretapping’s Fuzzy Future”. www.forbes.com. http://www.forbes.com/2008/05/15/wiretapping-voip-lichtblau-tech-security08-cx_ag_0515wiretap.html/. Retrieved 2009-03-02. 

^ Greenberg, Andy (May 15, 2008). “The State Of Cybersecurity  Wiretapping’s Fuzzy Future”. www.forbes.com. http://www.forbes.com/2008/05/15/wiretapping-voip-lichtblau-tech-security08-cx_ag_0515wiretap.html/. Retrieved 2009-03-02. 

^ GPO.gov, 47 C.F.R. pt. 9 (2007)

^ FCC.gov

^ Proenza, Francisco J.. “The Road to Broadband Development in Developing Countries is through Competition Driven by Wireless and VoIP” (PDF). http://www.e-forall.org/pdf/Wireless&VoIP;_10July2006.pdf. Retrieved 2008-04-07. 

^ Stripes.com, Stars and Stripes: USFK deal keeps VoIP access for troops

External links

Voice over Internet Protocol at the Open Directory Project

v  d  e

Computer-mediated communication

Online chat  Online discussion  Communication software  Collaborative software  Social network service

Asynchronous conferencing

Blog  E-mail  Electronic mailing list  FidoNet  Usenet  Internet forum  Shoutbox  Wiki

Synchronous conferencing

Data conferencing  Instant messaging  LAN messenger  Talker  Videoconferencing  Voice chat  VoIP  Web chat  Web conferencing

Categories: VoIP terminology & concepts | Broadband | Voice over IP | Telecommunications termsHidden categories: Articles needing additional references from January 2009 | All articles needing additional references | Law articles needing expert attention | Articles needing expert attention from November 2008 | All articles needing expert attention | All articles with unsourced statements | Articles with unsourced statements from December 2009 | Articles with unsourced statements from January 2009 | Articles needing cleanup from July 2009 | All pages needing cleanup | Articles with unsourced statements from March 2009

Defining IP telephony with cisco says:
IP telephony provides a way for you to extend consistent communications services to all your employees in their workspaces, whether they are in main campus locations, at branch offices, working remotely, or are mobile.IP telephony transmits voice communications over the network using open-standards-based Internet Protocol.
Cisco IP telephony solutions are an integral part of Cisco Unified Communications, which unify voice, video, data, and mobile applications on fixed and mobile networks enabling users to easily communicate in any workspace using any media, device, or operating system. Using the network as the platform, Cisco IP telephony solutions help organizations of all sizes realize greater security, resilience and scalability in addition to the inherent benefits of using a converged network for transport and interconnection.”
General Introduction of IP telephony or VoIPVoice over Internet Protocol (VoIP) is a general term for a family of transfer of technology for the provision of voice communication over IP networks like the Internet or other packet switched network. Other terms frequently encountered and is synonymous with VoIP IP Telephony, Internet telephony, Voice over Broadband (VoBB), broadband telephony, broadband and telephone.
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For you to install IP Telephony service, you would required following things:
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How VoIP calls works?

When you place a VoIP call your voice is converted into data packets which are routed through network of routers, switches and computers, when data packet stream reaches the destination it is then again converted into plain voice. That is how you a VoIP call is transmitted over IP networks.

How to place a Free VoIP call?

There are several methods to place a free VoIP call using certain technologies. Such as SIP (Session Initiation Protocol), SIP protocol is widely used by VoIP providers. You can use a computer or a simple VoIP device or an ATA (Analogue Telephone Adapter) to place a free VoIP call. You will need to register your VoIP device with a SIP service provider, there are number of freely available sip proxy in the market.

What features are available by using VoIP service?

All the features which are normally available on normal PSTN (Public Switch Telephone Network) line including advance features can be utilized in VoIP technology. Following is a standard list of features available.

Call waiting Three-way calling Call forwarding 69 (find out who last called you) 67 (block your number from caller ID) Repeat dialing (keeps trying a busy number) Call Recording Call Conference Video Call Conference

Can I use VoIP on Dialup internet connection?

The answer is yes, standard dialup connection is capable of providing 56kbps of data throughput, you can use G.723 codec which only utilizes 6 Kbps of bandwidth for crystal clear VoIP call, otherwise you can use G.729a/b/c codec which only utilizes 8 Kbps of bandwidth for crystal clear VoIP call. However Dialup internet connection for VoIP use is not recommended. Please be aware if you using any other wide band codec such as G.711a/u, Speex or GSM, the VoIP call quality will be degraded and highly not recommended.

Can I use VoIP on Wireless internet connection?

For a good quality VoIP call a minimum of 400 Milliseconds round trip latency is required, some wireless internet service providers latency is more than 600 milliseconds. If your wireless internet service provider’s network latency is less than 400ms then certainly you can use VoIP on wireless internet connection. But it is highly recommended that you use stable fixed line internet connection for high quality VoIP call.

Can I use FAX over VoIP?

The answer is yes you can attach your fax machine with VoIP ATA, but make sure that your service provider support T.38 / T.39 FAX protocols over SIP or IAX technologies.

Can I use VoIP line with Dialup Modem?

If you are using G.711 or any other wideband VoIP codec with less than 200ms latency then you can use VoIP line with Dialup Modem.

Can I send SMS through VoIP?

The answer is no, you cannot send Short Text Message (SMS) by using VoIP. However, there are quite a few companies online which lets you send SMS message by using internet.

Can I transfer my existing telephone number to VoIP line?

It depends where you are located, most companies in Europe and US will let you port your number to any service provider.

Can I use both PSTN and VoIP lines in single device?

Yes you can use both PSTN and VoIP lines in a single device, there are quite a few VoIP devices available in the market which lets you use both technologies.

Can I use my mobile phone to place VoIP calls?

Mobile phones are minimum computers and they contain same infrastructure as computers do, some mobile phones are equipped with SIP dialer or VoIP SIP clients which can be used to place mobile VoIP phone calls.

Can I call someone who does not have VoIP?

Yes you can call any number in the world, almost all VoIP providers will let you place a PSTN call for little charge.

Can I place 911 calls using VoIP?

Although 911 calls made via VoIP are unreliable. Also, emergency services  will not be able to track your location, as when you are using a regular phone. If you decide to replace your phone line with VoIP, consider using your cellular phone instead of your VoIP to call 911.

Do I have use a computer to place VoIP phone call?

It is not necessary to use a computer to place a VoIP call. Consider buying a VoIP device rather than using a computer to make and receive VoIP calls.

Why should I switch to VoIP?

VoIP technology is considerably costs less than traditional PSTN technologies, you can make and receive free phone calls and it definitely provides hundreds of more features than a traditional PSTN line.

In recent years Session Initiation Protocol or SIP has become a widespread protocol of choice for multimedia applications. In particular SIP has become quite popular in voice applications. Most major PBX manufacturers now support IP endpoints and also support SIP trunks and extensions. The growing popularity of the open source PBX software Asterisk and its various derivatives (e.g.Asterisk Business Edition, AsteriskNow, Elastix, Trixbox, Switchbox etc.) has also fueled the development and implementation of SIP.

Of note is the growth of the SIP compliant hardware industry. Major manufacturers like Aastra, Grandstream, Polycom, and Snom produce SIP phones that are in common use. Even proprietary PBX manufacturers like Avaya, Nortel, and Cisco have their own flavors of SIP firmware available for their IP telephones.

Polycom’s SoundPoint IP is one of the most popular lines of SIP compliant phones in today’s market. Not only limited to Asterisk based phone systems on the customer premise, these phones have also become ubiquitous in the hosted IP PBX market. The incorporation of a high-quality speakerphone, robust features, and customizable configuration via XML-based server files has laid the foundation for their implementation.

The prevalence of SoundPoint IP telephones has produced a burgeoning secondary market for the earlier models in the SoundPoint IP product line. Still well supported in both parts and firmware, the SoundPoint IP 300, 301, 430, 500, 501, 600, and 601 are great alternatives for the cost-conscious small business. The most popular in the secondary market are refurbished SoundPoint IP 430, 600, and 601’s. Featuring dual switched Ethernet ports for collocation of a PC and integrated 802.3af PoE support, these three models have made it easy to setup a workstation. No need for an auxiliary power supply or local AC power outlet. Using PoE, one LAN cable can run both the telephone and computer’s data connection.

What do you give up by choosing these models? Currently, most businesses wouldn’t give up too much. The most touted feature in the newer SoundPoint IP 450, 550, and 650 models is Polycom’s HD Voice technology. HD Voice is the moniker being given to wideband audio based on the G.722 codec. There are some early adopters reaping local benefits, but until there is widespread use, most small businesses will not notice the sound quality difference. Truth be told, the audio quality of the SoundPoint IP family is great as it is.

What do you gain by choosing these models? A solid and well-supported SIP compliant IP phone and a fatter wallet. Polycom continues to support its discontinued product for three years after end of sale. That means hardware and software support until 2013. By then, the HD Voice models should have a great secondary market as well.

What kind of savings will you realize? The price of a refurbished SoundPoint IP telephone can range from $69 (2-line) to $139 (6-line), whereas the newer models range from $179 (3-line) to $269 (6-line).

A Voice over Internet Protocol (VoIP) phone system requires the use of VoIP phones and there are several different types from which to choose. There are software based Session Initiation Protocol (SIP)/VoIP phones, USB VoIP phones, and hardware SIP phones. This is the new market for phone service. Providers of both telephone and Internet service have begun to offer their customers the opportunity to experience the new VoIP phone systems. They offer competitive packages and most have plans that give you the choice between limited and unlimited calling in the United States and Canada along with low International rates.

Software based VoIP phones require that you download a program to your computer. You can then use the downloaded soft phone program along with your computers microphone and speakers to make and receive calls. You can also use a headset with microphone if you prefer. This is a great choice if you use your computer for the majority of your communications. The software is easy to install and integrates well with other applications such as Microsoft Outlook.

USB VoIP phones plug into a port on your computer. You can then use them with VoIP software. They look and act like your traditional phone the difference being that the voice signal goes through your broadband service and not the Public Service Telephone Network (PSTN).

Hardware based SIP phones looks like your traditional phone. The difference being that they connect directly to your broadband network. These phones have a mini hub, which enables them to share the network connection with your computer.

You can use your current phone with VoIP services and connect it to the network via an adapter. The adapter uses an Ethernet cable that you use to plug the phone into the adapter and the adapter into the computer. You will find that setting up a VoIP phone is neither difficult nor expensive. It is highly recommended that you have a high-speed Internet connection for the service to work effectively.

IP phones look just like standard phones with a handset, cradle, and buttons and come with the standard connectors that you are accustomed to seeing. They also include an Ethernet connection as well. The purpose for this is to allow the IP phones to connect directly with you computer or router. They include the hardware and software required to handle IP calls. If you prefer to use your computer directly, you can also use VoIP for computer-to-computer calling. Certainly, this is the simplest method but you may experience a lower voice quality than with using an adapter or IP phone.

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This second edition handbook has been revamped to cover the newest standards, services, and products. You’ll find the latest on SIP usage beyond VoIP, including Presence, instant messaging (IM), mobility, and emergency services, as well as peer-to-peer SIP applications, quality-of-service, and security issues–everything you need to build and deploy today’s SIP services.

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