Internet Telephones Voice-over-Internet protocol: Voice-over-Internet protocol (VoIP) is a protocol optimized for the transmission of voice through the Internet or other packet-switched networks. VoIP is often used abstractly to refer to the actual transmission of voice (rather than the protocol implementing it). This latter concept is also referred to as IP telephony, Internet telephony, voice over broadband, broadband telephony, and broadband phone. VoIP providers may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET providers. Some cost savings are due to utilizing a single network to carry voice and data, especially where users have underused network capacity that can carry VoIP at no additional cost. VoIP-to-VoIP phone calls are sometimes free, while VoIP calls connecting to public switched telephone networks (VoIP-to-PSTN) may have a cost that is borne by the VoIP user. Voice-over-IP systems carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques, encapsulated in a data-packet stream over IP. There are two types of PSTN-to-VoIP services: Direct inward dialing (DID) and access numbers. DID will connect a caller directly to the VoIP user, while access numbers require the caller to provide an extension number for the called VoIP user. VoIP History: Voice-over-Internet Protocol has been a subject of interest almost since the first computer network. By 1973, voice was being transmitted over the early Internet (Arpanet). The technology for transmitting voice conversations over the Internet has been available to end-users since at least the early 1980s. In 1996, a shrink-wrapped software product called VocalTec Internet Phone (release 4) provided VoIP along with extra features such as voice mail and caller ID. However, it did not offer a gateway to the PSTN, so it was only possible to speak to other Vocaltec Internet Phone users. In 1997, Level 3 began development of its first softswitch (a term they invented in 1998); softswitches were designed to replace traditional hardware telephone switches by serving as gateways between telephone networks. Revenue in the VoIP industry in the US is set to grow by 24.3% in 2008 to $3.19 billion. Subscriber growth will drive revenue in the VoIP sector, with numbers expected to rise by 21.2% in 2008 to 16.6 million. VoIP Functionality: VoIP can facilitate tasks and provide services that may be more difficult to implement or more expensive 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. Conference calling, call forwarding, automatic redial, and caller ID; zero- or near-zero-cost features that traditional telecommunication companies (telcos) normally charge extra for. 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. Advanced Telephony features such as call routing, screen pops, and IVR implementations are easier and cheaper to implement and integrate. The fact that the phone call is on the same data network as a user's PC opens a new door to possibilities. Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (QoS) guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round-trip propagation delay (400–600 milliseconds for links through geostationary satellites). The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. This function is usually accomplished by means of a jitter buffer in the voice engine. Another challenge is routing VoIP traffic through firewalls and address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from 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 firewalls involve using protocols such as STUN or ICE. VoIP challenges: Available bandwidth Network Latency Packet loss Jitter Echo Security Reliability In rare cases, decoding of pulse dialing Many VoIP providers do not decode pulse dialing from older phones. The VoIP user may use a pulse-to-tone converter, if needed. Fixed delays cannot be controlled but some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, Diffserv). The principal cause of packet loss is congestion, which can sometimes be managed or avoided. Carrier VoIP networks avoid congestion by means of teletraffic engineering. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing audio, although this increases delay. This avoids a condition known as buffer underrun, in which the voice engine is missing audio since the next voice packet has not yet arrived. Common causes of echo include impedance mismatches in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end. Mass-market telephony: A major development starting in 2004 has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. Full phone service VoIP phone companies provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited calling to the U.S., and some to Canada or selected countries in Europe or Asia as well, for a flat monthly fee. These services take a wide variety of forms which can be more or less similar to traditional POTS. At one extreme, an analog telephone adapter (ATA) may be connected to the broadband Internet connection and an existing telephone jack in order to provide service nearly indistinguishable from POTS on all the other jacks in the residence. This type of service, which is fixed to one location, is generally offered by broadband Internet providers such as cable companies and telephone companies as a cheaper flat-rate traditional phone service. Often the phrase "VoIP" is not used in selling these services, but instead the industry has marketed the phrases "Internet Phone", "Digital Phone" or "Softphone" which is aimed at typical phone users who are not necessarily tech-savvy. Typically, the provider touts the advantage of being able to keep one's existing phone number. At the other extreme are services like Gizmo Project and Skype which rely on a software client on the computer in order to place a call over the network, where one user ID can be used on many different computers or in different locations on a laptop. In the middle lie services which also provide a telephone adapter for connecting to the broadband connection similar to the services offered by broadband providers (and in some cases also allow direct connections of SIP phones) but which are aimed at a more tech-savvy user and allow portability from location to location. One advantage of these two types of services is the ability to make and receive calls as one would at home, anywhere in the world, at no extra cost. No additional charges are incurred, as call diversion via the PSTN would, and the called party does not have to pay for the call. For example, if a subscriber with a home phone number in the U.S. or Canada calls someone else within his local calling area, it will be treated as a local call regardless of where that person is in the world. Often the user may elect to use someone else's area code as his own to minimize phone costs to a frequently called long-distance number. For some users, the broadband phone complements, rather than replaces, a PSTN line, due to a number of inconveniences compared to traditional services. VoIP requires a broadband Internet connection and, if a telephone adapter is used, a power adapter is usually needed. In the case of a power failure, VoIP services will generally not function. Additionally, a call to an emergency services number may not automatically be routed to the nearest local emergency dispatch center. Some VoIP providers only handle emergency call for one country. Some VoIP providers offer users the ability to register their address so that emergency services work as expected. Another challenge for these services is the proper handling of outgoing calls from fax machines, DVR boxes, satellite television receivers, alarm systems, conventional modems or FAXmodems, and other similar devices that depend on access to a voice-grade telephone line for some or all of their functionality. At present, these types of calls sometimes go through without any problems, but in other cases they will not go through at all. And in some cases, this equipment can be made to work over a VoIP connection if the sending speed can be changed to a lower bits per second rate. If VoIP and cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional voice-grade telephone line would be available in almost all homes in North America and western Europe. The TestYourVoIP Web site offers a free service to test the quality of or diagnose an Internet connection by placing simulated VoIP calls from any Java-enabled Web browser, or from any phone or VoIP device capable of calling the PSTN. Mobile Number Portability (MNP) in the Internet Telephony Environment: Mobile number portability (MNP) also impacts the internet telephony, or VoIP (Voice over IP) business. A voice call originated in the VoIP environment which is routed to a mobile phone number of a traditional mobile carrier also face challenges to reach its destination in case the mobile phone number is ported. Mobile number portability is a service that makes it possible for subscribers to keep their existing mobile phone number when changing the service provider (or mobile operator). VoIP is clearly identified as a Least Cost Routing (LCR) voice routing 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. 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 now need to know the actual current 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 the home network a mobile phone number belongs to. As VoIP starts to take off 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 assuring that the voice call will actually work, VoIP companies give businesses the necessary reliability they look for in an internet telephony provider. UK-based messaging operator Tyntec provides a Voice Network Query service, which helps not only traditional voice carriers but also VoIP providers to query the GSM network to find out the home network of a ported number. 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. 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.

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