VoIP Implementations

Voice over IP has been implemented in various ways using both proprietary and open protocols and standards. Examples of available VoIP implementations include:

• SIP/RTP
• IMS
• H.323
• Skype

Further examples and comparisons are available from the following Wikipedia article: Comparison of VoIP software

Adoption

Consumer market

A major development starting in 2004 has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. Full phone service VoIP phone companies provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited calling to the U.S., and some to Canada or selected countries in Europe or Asia as well, for a flat monthly fee 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:

• 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 therefore most people also 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. An advantage of using a softphone with a VoIP service provider is the ability of having a fixed phone number which you can move to any country or location (This is also possible with ATAs and VoIP phones, however requires the physical relocation of the 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 slowly 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 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 main sets of competitors are fighting it out-- one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.

VoIP also offers the advantage of running both voice and data communications over a single network which can represent a significant saving in infrastructure costs.

Other advantages that appeal to business is that the per extension prices of VoIP are lower than those of PBXs or key systems. Also, VoIP switches rely on commodity hardware, such as PCs or Linux systems, so they are easy to configure and troubleshoot. Rather than closed architectures, these devices rely on standard interfaces.

VoIP devices also have simple, intuitive user interfaces, so employees can often make simple system configuration changes. Features such as dual-mode cellphones enable users to continue their conversations as they move from an outside cellular service to an internal wi-fi network. The bundling means employees no longer have to carry a desktop phone and a cellphone, so companies can reduce their telecommunications equipment costs. Maintenance also becomes simpler, because there are fewer devices to oversee.

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 for free from open source VoIP implementations such as Asterisk.

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 the same broadband connection. This can make VoIP a simple way to add an extra telephone line to a home or office.
• Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.) Most of the difficulties of creating a secure phone connection over traditional phone lines, like 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 an Internet connection is needed to get a connection to a VoIP provider. For instance, call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection.
• Integration with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books, and passing information about whether others (e.g., friends or colleagues) are available to interested parties.

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[citation needed]. 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.

In countries such as Singapore, the most recent Mobile number portability solution is expected to open the doors to new business opportunities for non-traditional telecommunication service providers like wireless broadband providers and voice over IP (VoIP) providers[citation needed].

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.

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 an offence in the USA to "knowingly transmit misleading or inaccurate caller identification information with the intent to defraud, cause harm, or wrongfully obtain anything of value ...".

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 to work like a traditional fax machine and can work using several configurations. 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. [46] Originally, T.38 was designed to use UDP and TCP transmission methods across an IP network. The main difference between using UDP and TCP methods for a FAX is the real time streaming attributes. 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[citation needed].

There have been updated versions of T.30 to resolve the fax over IP issues, which is the core fax protocol. Some new fax machines have T.38 built-in capabilities which allow the user to plug right into the network with minimal configuration changes[citation needed]. A unique feature of T.38 is that each packet contains a copy of the main data in the previous packet. This is an option and most implementations seem to support it. This forward error correction scheme makes T.38 far more tolerant of dropped packets than VoIP[citation needed]. 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 a high probability that you will receive the whole transmission.

Tweaking the settings on the T.30 and T.38 protocols could also turn your unreliable fax into a robust machine[citation needed]. Some fax machines pause at the end of a line to allow the paper feed to catch up. This is good news for packets that were lost or delayed because it gives them a chance to catch up. However, were this to happen on every line, your fax transmittal would take a long time. Another possible solution is to treat the fax system as a message switching system, which does not need a real-time data transmission (such as sending a fax as an email attachment (see Fax) or remote printout (see Internet Printing Protocol)). The end system can completely buffer the incoming fax data before displaying or printing the fax image.