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Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.

Internet telephony refers to communications services — voice, facsimile, and/or voice-messaging applications — that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.[1]

VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. Codec use is varied between different implementations of VoIP (and often a range of codecs are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs.

Contents

History

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:

The Session Initiation Protocol has gained widespread 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[11] 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.[12] 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".[13][14]

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.[15]

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.[16] 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.[17]

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.[18]

VoIP solutions aimed at businesses have evolved into "unified communications" services that treat all communications—phone calls, faxes, voice mail, e-mail, Web conferences and more—as 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.[19]

VoIP runs both voice and data communications over a single network, which can significantly reduce infrastructure costs.[20]

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.[20]

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.[20]

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.[21]

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.[22][23]

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.[24]
  • 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[25] 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.[26]
  • 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

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[27] 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 receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. Jitter results from the rapid and random (i.e., unpredictable) changes in queue lengths along a given Internet path due to competition from other users for the same transmission links. VoIP receivers counter jitter by storing incoming packets briefly in a "de-jitter" or "playout" buffer, deliberately increasing latency to increase the chance that each packet will be on hand when it's time for the voice engine to play it. The added delay is thus a compromise between excessive latency and excessive dropout, i.e., momentary audio interruptions.

Although jitter is a random variable, it is the sum of several other random variables that are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. Thus according to the central limit theorem, we can model jitter as a gaussian random variable. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful. In practice, however, the variance in latency of many Internet paths is dominated by a small number (often one) relatively slow and congested "bottleneck" link(s). Most Internet backbone links are now so fast (e.g., 10 Gb/s) that their delays are dominated by the transmission medium (i.e., 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 and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing).[28] 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.[29] 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.[30]

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 number—clearly 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.[31] 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.[32]

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.[33] 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.[34] 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.[35] VoIP implementations can also allow other identification techniques to be used. For example, Skype allows subscribers to choose "Skype names"[36] (usernames) whereas SIP implementations can use URIs[37] 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[38] and the ENUM service in IMS and SIP.[39]

Echo can also be an issue for PSTN integration[40] . Common causes of echo include impedance mismatches in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end.

Security

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.[41]

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.[42] 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.[43]

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.[44] 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.[45][46][47][48][49] Public Secure VoIP is also available with free GNU programs.[50]

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[51] 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 ..."[52]

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.[53]

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.[54] 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.[55] 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.[56]

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.[57] 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.[58]

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 warrant—but 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.[59] 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.[60] 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.[61]

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.[62] 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.[63]

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.[64] 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 the popularity of 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.[65]

International VoIP implementation

IP telephony in Japan

In Japan, IP telephony (IP電話 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

References

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  2. ^ 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
  3. ^ "The Launch of NSFNET". The National Science Foundation. http://www.nsf.gov/about/history/nsf0050/internet/launch.htm. Retrieved 2009-01-21. 
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  7. ^ "RFC 2235". R. Zakon. http://www.faqs.org/rfcs/rfc2235.html. Retrieved 2009-01-21. 
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  16. ^ Vodafone Terms and Conditions & Mobile Phones from Phones 4u
  17. ^ NEWS.BBC.co.uk, "T-Mobile must open Truphone lines" — BBC
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  20. ^ 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. 
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  29. ^ ICT Regulation Tool Kit — 4.4 VoIP — Regulatory Issues — Universal Service
  30. ^ Letter from the City of New York to the Federal Communications Commission
  31. ^ FCC Consumer Advisory: "VoIP and 911 Service"
  32. ^ 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.
  33. ^ Keeping your telephone number when you change your service provider — FCC
  34. ^ VoIpservice.com
  35. ^ "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. 
  36. ^ "Create a Skype Name". Skype. http://www.skype.com/help/guides/createskypename_windows/. Retrieved 2009-01-21. 
  37. ^ "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. 
  38. ^ "Your personal online number". Skype. http://www.skype.com/allfeatures/onlinenumber/. Retrieved 2009-01-21. 
  39. ^ "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. 
  40. ^ Packetcable Implementation P557 — Jeff Riddel — ISBN 1587051818 Google Books Preview
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  42. ^ "Examining Two Well-Known Attacks on VoIP". CircleID. http://www.circleid.com/posts/examining_two_well_known_attacks_on_voip1/. Retrieved 2006-04-05. 
  43. ^ Skyp.com, "Skype Security Evaluation", Tom Berson/Anagram Laboratories
  44. ^ Disa.mil, Internet Protocol Telephony & Voice over Internet Protocol Security Technical Implementation Guide
  45. ^ IJCSNS.org, Secure Voice-over-IP
  46. ^ Sans.org, SANS Institute InfoSec Reading Room
  47. ^ JHU.edu, Packet Loss Concealment in a Secure Voice over IP Environment
  48. ^ GDC4S.com, State-of-the-art voice over IP encryptor
  49. ^ Networkworld.com, Cellcrypt secure VoIP heading to BlackBerry.
  50. ^ Freesorftwaremagazine.com, Secure VoIP calling, free software, and the right to privacy
  51. ^ VoIPSA.org, Blog: "Hello Mom, I'm a Fake!" (Telespoof and Fakecaller).
  52. ^ Govtrack.us, "Truth in Caller ID Act".
  53. ^ Oldphoneworks.com
  54. ^ Soft-Switch.org, Faxing over IP networks
  55. ^ Umass.edu, UMass Discussion on UDP transmission Characteristics.
  56. ^ Faqs.org, RFC 3362-T.38
  57. ^ FaxSuperstore.com, Ricoh 4410NF
  58. ^ "Global VoIP Policy Status Matrix". Global IP Alliance. http://www.ipall.org/matrix/. Retrieved 2006-11-23. 
  59. ^ 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. 
  60. ^ 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. 
  61. ^ 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. 
  62. ^ GPO.gov, 47 C.F.R. pt. 9 (2007)
  63. ^ FCC.gov
  64. ^ 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. 
  65. ^ Stripes.com, Stars and Stripes: USFK deal keeps VoIP access for troops

External links


Wikibooks

Up to date as of January 23, 2010

From Wikibooks, the open-content textbooks collection

Voice over IP

This book is a work in progress. Please feel free to add, correct or provide comments/criticism via the Discussion page.

Contents

Introduction

Voice over IP (or VoIP) is an emerging technology allowing encapsulation of voice telephony services within IP (Internet Protocol) for transmission over IP networks, such as the Internet.

VoIP technology provides cheap and flexible voice telephony by utilizing packet-switched networks for voice data, providing a much cheaper solution than traditional circuit-switched telecommunications services, while also allowing termination of IP calls closer to the intended destination, avoiding long-distance tolls.

With the global prevalence of broadband technology at home, interest in telecommutation technology and a move away from traditional copper line telephony, there is a lot of industry interest in VoIP technology.

Purpose

This book is intended for those who possess moderate to high computing skills. The initial sections of the book, dealing with VoIP history and simple VoIP deployment for home and small business users, is specifically written to allow those interested in VoIP technology to get an opportunity to try it out without getting too technical. For those more technically inclined, subjects such as VoIP protocol analysis and advanced PBX functions are discussed toward the end of the book.

Moving away from traditional telephony

There are many compelling reasons for individuals and businesses to invest in Voice over IP. For smaller businesses or home users who frequently make long distance or timed phone calls, to larger commercial ventures looking to reduce costs on site-to-site telephony.

The range of VoIP plans available also allow more flexibility and cost control. Because VoIP termination does not involve any physical rental or consume resources when idle, plans are available on a per-use or prepaid basis, on a monthly subscription basis, or in any number of other configurations.

A successful VoIP deployment requires good planning and a good understanding of the strengths and weaknesses of IP Telephony. Using the right hardware, the right VoIP provider(s), and a customised dialplan can save significant amounts of money.

Both the positive and negative aspects of VoIP technology are detailed in the following sections.

Benefits of VoIP

Cheaper long-distance - Because VoIP calls can be terminated anywhere in the world, long-distance calls can be made for much cheaper rates. For example, if you are a European VoIP user who frequently calls US numbers, you can purchase VoIP termination credit from a US VoIP provider, which will generally offer near local call prices from anywhere in the world.

Incoming Calls - For the same reason as above, VoIP provides the ability to obtain incoming telephone numbers from anywhere around the world, routed to you over the Internet. It's possible to obtain US 1-800 numbers, for example, no matter what your location.

No copper rental - Without the need for a circuit from your residence or business to your telephone company's exchange, your monthly costs can reduce significantly. Yet, there must be some means of connectivity from the "End User A" who initiates a call onto the Internet. Ostensibly, this connection is either a broadband connection (Cable or Telco DSL), or a "dial-up" connection. In both cases, and in this sense, obviously a copper rental is required.

Dial-Plan freedom - Using a dial plan, you have the ability to direct your softphone, hardware phone or VoIP gateway to send calls to different VoIP gateways or via land-line, depending on the number dialled. You also have the ability to set priority numbers, directing the gateway to drop an existing call in favour of the priority call where no lines are available.

Free IP-to-IP Calls - Where both the caller and recipient are using VoIP to place/receive calls, and where the correct configuration exists (see the section on ENUM at the end of this book), calls can be placed entirely over the Internet without any charge.

VoIP Issues

As with any new technology, Voice over IP has its share of issues which must be considered carefully before implementation.

Emergency and Information services - There is no guarantee that calls terminated over IP for numbers such as 911 and 411 (in the United States) will be correctly routed. Due to the lack of location-specific termination of Voice over IP calls, it is not possible for such services to determine your location, and it is possible that the gateway your calls are terminating on does not have access to such numbers at all.

This limitation, and suggestions for workarounds, are discussed later in this book.

Bandwidth Issues - VoIP protocols can be very sensitive to both bandwidth limitations and network latency. Generally, VoIP calls must be placed via a broadband connection. Where there is heavy bandwidth utilization, high network latency, or the gateway is a significant distance from the user, call quality can be heavily degraded with the possibility that there will be no VoIP service at all.

See the chapter entitled traffic shaping and bandwidth control later in this book for suggestions on how this can be controlled.

NAT/PAT Issues - NAT (Network Address Translation) and PAT (Port Address Translation) are used to provide connectivity to machines in many situations such as private networks which require Internet connectivity.

With NAT, a gateway/router system will translate an IP address from the source address provided by the machine into another IP address entirely. NAT is often confused with PAT; however NAT provides a 1:1 relationship, eg. 192.168.10.10 will be translated to 203.10.10.10.

PAT is a slightly different translation. With PAT, one or more public IP addresses exist for any number of private/translated addresses. In the most common setup, a home user may have a residential DSL connection and a single public IP address, however they may have several computers on a private network behind their gateway/router which require Internet connectivity.

With PAT, the router/gateway will translate all private IP addresses into the public address provided, and will keep track of which ports established via this address belong to which private IP addresses, allowing the machines to share the public IP address.

VoIP protocols often have inherent issues with NAT and PAT. Due to the prevalence of these gateway devices for cable, wireless and DSL users, even users with single machines may find that PAT is performed by default via their router.

Please see the section immediately below entitled VoIP Protocols for more information on which protocols in particular are affected by NAT and PAT and how to avoid problems.

Standard Architectures for Next Generation Networks

IMS(IP Multimedia Subsystem)

IMS is currently in development and testing phases, and was originally developed by 3GPP as an architecture for Mobile Networks.

IMS describes a full architecture that uses IP technologies to transport all traffic (both Voice and Data). It tries to re-use existing protocols defined by IETF and other standards bodies instead of re-inventing the wheel. Some of the protocols it reuses are SIP, MEGACO/H.248, SIGTRAN and DIAMETER.

TISPAN

TISPAN reuses as much as possible from the IMS architecture and complements it with the required elements to serve fixed/wireline networks.

The work developed by TISPAN is then feed back into IMS.

Voice over IP Technology

The term VoIP is often used to describe general IP telephony technology. Behind this term lies a range of different protocols, languages used to communicate between a range of different devices and vendors.

These protocols are one of the most important factors in choosing a VoIP solution. It is generally impossible to get different devices to communicate, unless they support the same protocol.

VoIP Protocols

The following protocols are associated with VoIP voice call termination and inter-gateway communication.

H.323 - H.323 is used by many commercial vendors for IP telephony, It is a suite of protocols which includes H.245, Q.931, etc. This suite takes care of session establishment between phone and switch and is heavily based on the Integrated Services Digital Network (ISDN) signalling protocols. Signalling part is also taken care by this protocol. This suite uses RTP, RTCP, RSVP for actual data transfer and QOS between phones.

SIP (Session Initiation Protocol) - SIP is a standard developed by the IETF (Internet Engineering Task Force) for use in establishing multimedia sessions such as voice, instant messaging and video, amongst other applications.

SIP is also responsible for the implementation of other voice session related functions such as holding voice calls, transferring calls, or hosting several voice conversations simultaneously (call conferencing).

SIP utilises the Session Description Protocol (SDP) to negotiate data types available at either end. Voice and video data carrying is typically performed with the RTP and RTCP protocols.

MGCP (Media Gateway Control Protocol) - MGCP is a protocol typically used internally to systems to represent the whole system as a single entity. MGCP systems are made up of Call Agents and Gateways. The Call Agents keep stateful information and offload the majority of the control work from the Gateway(s). Voice and video carrying is performed by RTP and RTCP.

MEGACO/H.248 - MEGACO is very similar to MGCP and is the standard used for Call Agent architectures.

IAX (Inter Asterisk eXchange) - The IAX protocol was developed by a team of open-source developers working on the Asterisk project, a very popular and successful open-source PBX described later in this document.

There is not much industry support for IAX, however support has been slowly increasing. There has been a lack of good documentation for the IAX protocol, and many commercial vendors are hesitant to give their support to a protocol not ratified by a standards body such as the IETF or ITU.

LTP (Lightweight Telephony Protocol) - the LTP is a binary lightweight protocol that is NAT friendly and based entirely on free codecs. It is easily understood and in use since 1999.

VoIP Phone Hardware/Software

VoIP hardware and software implementations allow phone calls to be placed over IP. Software which uses a computer's soundcard device to provide voice input and output is known as a Softphone.

Several hardware devices exist to provide VoIP telephony. ATA devices provide standard phone ports, allowing standard phone handsets and other devices such as fax machines to utilize VoIP services. Specialised VoIP handsets are also available, often providing advanced features supported by VoIP gateways.

Hardware phones generally provide a better quality VoIP experience than Softphones; however the convenience of not requiring extra hardware, and the smaller expense involved in Softphone products make softphones a popular choice.

Softphones

A softphone is a soundcard device to provide voice input (via microphone) and sound output (via headphones/speakers) without the need for extra hardware. Handsets and headsets are available which plug into the soundcard and resemble normal phone headsets/handsets, making softphone use more comfortable.

A popular open source softphone that runs on both Windows and Linux platforms is ekiga which supports both H.323 and SIP protocols. Xten X-Lite is another popular free Softphone which provides all necessary features to initiate and receive phone calls via SIP. A list of some of the softphones available can be found on voip-info.org here, and on Wikipedia here.

VoIP Handsets

ATA Adaptors

`ATA (Analog Telephone Adaptor) Devices provide the ability to connect standard telephone handsets to a hardware unit which provides VoIP capability.

An advantage of ATA devices is the ability to connect other telephony devices, such as TTY devices, FAX machines, Pay TV units and even modems.

VoIP Gateway Systems

This chapter deals with the complex world of VoIP gateway systems. These systems are responsible for providing some amount of call control and routing from VoIP phone devices.

Each of these systems is responsible for receiving calls from a VoIP handset, ATA or softphone device via a supported VoIP protocol, consulting a dial-plan or other call routing table, and correctly routing the call. Other features made available by some or all of these gateways include:

Authentication - Ensuring that clients have permission to use VoIP resources.

Call Accounting - Providing tracking of calls via the gateway, including reports and cost controls.

Complex Dialplans - Most dialplans configured on simple devices such as ATAs and softphones only allow for a very straightforward dialplan configuration. Using a VoIP gateway such as Asterisk, factors such as time of day, or even values obtained from external sources can be used to determine which path a call should take.

Hardware Integration - Most gateways provide the ability to interface with physical hardware such as FXS (internal line) and FXO (PSTN line) cards to create a PBX system. They additionally provide the dialplan configuration to determine which calls should route via IP, and which should utilize a card. It is important when selecting gateway software to evaluate which devices they support, and ensure yours is listed. Some devices (for example, Cisco line cards) may require the vendor's specific implementation of VoIP gateway for any support at all.

Internal Numbering - Some gateways offer the ability to provide internal extension numbering for clients, allowing one VoIP device to dial another device registered on your local gateway using a short internal extension number.

When would you use a VoIP gateway?

In which situations would you use a VoIP gateway product?

Generally, if you require an advanced configuration such as advanced call accounting for billing, cost controls, complex dialplans, or the ability to call between extensions, you should implement a gateway solution.

If you are interested in integrating extra hardware such as FXO cards (interfaces to the PSTN) you MUST use a gateway product.

If you are designing VoIP solutions for mid-large size business, or will be requiring IVR (Interactive Voice Response) functionality, you should implement a VoIP gateway.

Open-Source Gateways Commercial Gateways
Asterisk
Bayonne
OpenSER
SIP Express Router
Yate
YXA

Asterisk: The open-source PBX

Asterisk is an extremely popular open-source PBX system which runs on BSD, Linux, Mac OS X, and Windows. The project is sponsored by Digium, a PBX hardware manufacturer.

Asterisk has support for ENUM, e911, Caller ID, all call controls such as Forwarding, Conferencing, Hold, Transfer and Call Waiting. Additional features such as Call Monitoring, Call Recording and Privacy Controls also exist.

In addition to the above, Asterisk is able to provide IVR functions, allowing interactive voice prompts, call queuing, and many many advanced call routing features.

In all, Asterisk is a remarkably full featured commercial-grade PBX system available free of charge.

Bayonne

OpenSER

Web page: http://openser.org

OpenSER is a robust and powerful SIP server. Released under GPL, OpenSER is the first free server with integrated TLS, offering secure VoIP communications. It has an architecture designed for scalability and flexibility and high performances.

Main characteristics:

  • SIP proxy/registrar/redirect server (RFC3261)
  • transaction stateful
  • UDP/TCP/TLS support
  • modular architecture
  • scripting configuration file with pseudo-variables
  • authentication, authorization and accounting via database, radius or text files
  • enum support
  • NAT traversal system
  • formatted logging
  • least cost routing
  • Call Processing Language (CPL)
  • MySQL/Postgres/Flat files database backend
  • sever monitoring

SIP Express Router

Web page: http://www.iptel.org/ser

SIP Express Router (SERi) is a high-performance, configurable, free SIP server licensed under the open-source GNU license . It can act as SIP (RFC 3261) registrar, proxy or redirect server. SER can be configured to serve specialized purposes such as load balancing or SIP front-end to application servers, SEMS for example.

SER features:

  • complete support of RFC 3261 functionality,
  • a variety of database backends (mysql, oracle, postgres, radius, text-db),
  • management features (remote management via XML-RPC, load-balancing),
  • NATi traversal, telephony features (LCR, speeddial),
  • multidomain hosting, ENUM, presence, and even more.

SER is additionally enhanced by a variety of additional SIP tools, which provide functionality for management, media processing, CDRi processing, etc.

SER is today default part of numerous operating systems and their distributions: Debian, FreeBSD, Gentoo, NetBSD, OpenBSD, OpenSUSE, Solaris.

SER history spans back to the previous century. SER has been used since 2002 for various different purposes, frequently in the industry by major ISPs/ASPs and by universities to enable VoIPi services. SER's particular strength is its performance (SER runs well even under heavy load caused by large subscriber populations or abnormal operational conditions), flexibility (SER's genuine configuration language and module interface allow high degree of customization) and interoperability (tested and operated against tens of SIP products over the years, including but not limited to (Microsoft, Cisco, Mitel, snom, Pingtel, Siemens, xten, and many others).

Yate - Yet Another Telephony Engine

Yate is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP), its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.

Yate can be used as a:

  • VoIP server ****
  • VoIP client
  • VoIP to PSTN gateway
  • PC2Phone and Phone2PC gateway
  • H.323 gatekeeper
  • H.323 multiple endpoint server
  • SIP session border controller
  • SIP router
  • SIP registration server
  • IAX server and client
  • IP Telephony server and client
  • Call center server
  • IVR engine
  • Prepaid and post-paid cards system

YXA

YXA is a SIP server written in the programming language Erlang [1] at Kungliga Tekniska Högskolan and Stockholms universitet. Erlang was developed by Ericsson to program ordinary telephone switches, with the goal of making a programming system fault-tolerant and robust.

This helps YXA to be a robust SIP server/stack capable of serving tens of thousands of users. The projects goal is to make YXA complient to all RFC standards relevant to SIP.

  • It is RFC3261 compliant SIP-server, capable of everything a generic domain needs :
    • Registrar that keeps track of your users
    • Handles incoming SIP requests to your domain
    • Handles routing of requests from your users to remote domains
    • TCP, UDP and TLS (including SIPS) support
    • Automatically maps e-mail addresses of your users to their SIP addresses, if you have the e-mail addresses in LDAP
    • Handles multiple domains using a single server instance
  • ENUM support for PSTN-bypass whenever possible
  • IPv6 support
  • Forking, both parallel and sequential
  • CPL (RFC3880) support for advanced user-control of events (currently incoming calls only)
  • Modular user database, currently with LDAP, Mnesia, MySQL and text-file backends
  • PSTN destination access control (per user or for anonymous users)

VoIP PABX Integration

Advanced VoIP Configuration

Traffic shaping and bandwidth control

Correctly terminating emergency calls

Particular attention should be given to ensuring that emergency calls on VoIP lines are handled correctly. In some cases, it may not be possible to provide useful or reliable emergency service via VoIP services. In this case, it may be possible to offload emergency calls to a landline service.

If reliable emergency service is not available, the handset or VoIP device should clearly be marked with a warning.

Some VoIP providers allow emergency calls to be routed via their service; however this is only of use if you are within an area covered by the receiving emergency call centre. Additionally, a lack of bandwidth or Internet service or a power loss can lead to loss of emergency service via telephone.

Suggestions for dealing with emergency calls

If a functioning landline handset is nearby, it is advisable to attach a notice to the VoIP device directing emergency calls to the landline. It is much easier for emergency services to assist if they have information on your location which is generally not available via VoIP calls.

Where calls are handled by a gateway administered by yourself or your organisation, and the ability to route certain calls via landline exists, the dialplan for your VoIP gateway should immediately route emergency calls via the landline, and where possible, any active conversations should be dropped if there are not sufficient lines to handle the call.

It is important to find out whether your VoIP provider(s) offer emergency numbers, what areas they are able to cover and whether they are able to provide enhanced emergency services such as e911 initiatives which allow providers to transmit information on subscribers to 911 call centres.

Ensure that emergency calls are routed via the most appropriate gateways in your dial plan.

Incoming telecommuncations services

IVR (Interactive Voice Response) application

IVR applications are automated response units which receive input from the user (traditionally in the form of DTMF) and allow the user to progress along a pre-set interactive path. For example, a menuing system where the IVR may announce Press 1 for Sales, 2 for Marketting and then await user input. Once the user input arrives the IVR unit may patch through the call to another line and/or may provide a pre-recorded message.

ENUM/E164 Technology

The following section is quite technical, as it deals with complicated infrastructure and the integration of VoIP into core internet technologies.

The ENUM protocol (described in RFC document RFC 3761) translates standard telephone number routing (known as E.164, as the ITU defines the standard by which telephone numbers are assigned in the E.164 standard) into an addressing scheme able to route to VoIP (eg. SIP/H323) gateways using DNS technology.

ENUM technology allows VoIP servers to use the DNS infrastructure to query routing paths for terminating voice calls given a phone number.

When a call is initiated to a given phone number, a VoIP gateway capable of ENUM service will query the provided number in DNS to determine whether there is a path other than the PSTN line available to the destination service.

This allows providers to advertise paths via VoIP to services otherwise only available via a VoIP to PSTN gateway.

ENUM records may contain other information in addition to a VoIP gateway address, such as a web URL or e-mail address.

Private vs Public ENUM databases

The public ENUM database is a collection of ENUM records, known as NAPTR RRs in DNS speak, which map telephone numbers to VoIP service gateways. These records are stored in a hierarchical tree beneath the e164.arpa domain.

Records within the e164.arpa domain are delegated to their responsible owner via the E.164 addressing convention. Much like the in-addr.arpa domain, which is used to provide mapping of IP addresses to names, these records are delegated in blocks to the agency responsible for this number assignment.

The agency is then responsible for further smaller delegations where necessary. As a result, it is unlikely that individuals or small businesses will have access to provide ENUM records for their telephone numbers.

It also stands to reason that telephone companies would be hesitant to allow telephone calls destined to their customers to instead terminate over the Internet, as much of a phone company's revenue would be derived from call termination fees, charged to carriers for calls terminating on a company's network.

This is unfortunate for end users. Via the ENUM database, you would have the opportunity to direct VoIP-based calls to terminate over the Internet via your VoIP gateway, rather than via a phone line.

For this reason, several free ENUM databases have been created to allow users to share alternate IP-based paths to their phone numbers. Most ENUM aware gateways will allow additional 'search domains' to be configured, providing access to these unofficial databases.

Whilst configuring a gateway to query these extra domains can dramatically reduce costs by terminating calls over IP, therefore freeing copper lines/voice channels for both the caller and recipient, there is an important factor to consider. If a malicious or unauthenticated entry is added, an attacker can direct calls for a phone number to an arbitary VoIP gateway.

Consider the example of rival businesses. If Business A were tech-savvy and entered their competitor's number into a public directory, directing calls to their own VoIP gateway, any calls placed via a VoIP gateway configured to query that directory would be directed to Business A's phone, rather than the intended recipient, Business B.

Some of these directories require voice verification of ownership when you register phone numbers. Others do not require verification or have dispute resolution policies which deal with disputes on a case-by-case basis.

Another less serious but still important consideration is the possibility that your information will be harvested from the database, or sold by the database owner. ENUM information may be valuable to advertisers or phone spammers, as it would provide them with a list of valid phone numbers, and allow them to call for free via IP.

Public ENUM registries

Free Registries

e164.org - International E.164 database.

e164.dk - Danish E.164 database.

q.nemox.net - Austrian E.164 database. This was the first public ENUM database on the Internet.

SIP.edu Exchange - The SIP.edu exchange provides access to many educational VoIP gateways, for institutions such as MIT, UC Berkeley, UCLA, and others.

thevpf.com - The Voice Peering Fabric registry.

Commerical Registries

ENUM Trial Registries

Many ENUM trial registries are becoming available. Most of these registries are regional, and often are limited by telephony provider, limited number of trial participants, or a trial fee.

Australian ENUM Trial - Fee applies for registration

Austrian ENUM Trial - Open to Austrian residents, a fee applies for registration.

German ENUM Trial

Japanese ENUM Trial - Open to ETJP contributors since 2003. Currently has 46 members.

Taiwan ENUM Trial

UK ENUM Trial

VoIP Resources

  • www.voip-info.org - An excellent wiki resource for VoIP users interested in how to deploy, debug, administer and enhance VoIP systems.
  • VOIP Information A guide to VOIP (general)

Appendices

Appendix A: Glossary of Terms

ATA Analog Telephone Adaptor
Broadband Using multiple transmissions simultaneously over a single medium.
Baseband Using a single transmission over a single medium.
DTMF Dual-Tone Multi-Frequency
DSL Digital Subscriber Line
FXO Foreign eXchange Office
FXS Foreign eXchange Subscriber
IP Internet Protocol - IP is the common protocol, or language, used by all devices on the Internet to communicate regardless of vendor.
MGCP Media Gateway Control Protocol
NAT Network Address Translation - The translation of a Network (IP) Address to another address by a gateway or router device
PAT Port Address Translation - The translation of multiple inside IP addresses to a single IP address on the outside, used in typical home applications.
PBX Private Branch Exchange
POTS Plain Old Telephone System. See PSTN below.
PSTN Public Switched Telephone Network
RFC Request For Comments
RTCP Real-Time Control Protocol
RTP Real-Time Protocol
SIP Session Initiation Protocol

For further reading








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