Voice over Internet Protocol (VoIP)

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). VoIP is also known as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband. "VoIP" is pronounced voyp.

Companies providing VoIP service are commonly referred to as providers, and protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They 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 existing underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to public switched telephone networks, PSTN, may have a cost that is borne by the VoIP user.

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.^[1] The technology for transmitting voice conversations over the internet has been available to end users since at least the 1990's. 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 analog POTS, so it was only possible to speak to other Vocaltec Internet Phone users.^[2] In 1997, Level 3 began development of its first softswitch (a term they invented in 1998); softswitches were designed to replace a traditional hardware switchboards by serving as the gateway between two telephone networks.^[3]

Functionality

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

• The ability to transmit more than one telephone call down the same broadband-connected telephone line. This can make VoIP a simple way to add an extra telephone line to a home or office.
• 3-way calling, call forwarding, automatic redial, and caller ID; 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 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 online to interested parties.

Implementation

Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (known as 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 milliseconds to 600 milliseconds for geostationary satellite). 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 functionality is usually accomplished by means of a jitter buffer.

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 a protected enterprise network. 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
• Delay/Network Latency
• Packet loss
• Jitter
• Echo
• Security
• Reliability
• Pulse dialing to DTMF translation

Many VoIP providers do not translate pulse dialing from older phones to DTMF. The VoIP user may use a VoIP Pulse to Tone Converter, if needed.^{[citation 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 be controlled by congestion management and avoidance. 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 buffer (called a play-out buffer) upon arrival, before playing them out. This avoids a condition known as buffer underrun, in which the playout process runs out of voice data to play because the next voice packet has not yet arrived, but increases delay by the length of the buffer.

Common causes of echo include impedance mismatches in analog circuitry, and acoustic coupling of the transmit and receive signal at the receiving end.

Reliability

Conventional phones are connected directly to telephone company phone lines, which in the event of a power failure are kept functioning by back-up generators or batteries located at the telephone exchange. However, household VoIP hardware uses broadband modems and other equipment powered by household electricity, which may be subject to outages in the absence of a uninterruptible power supply or generator. Early adopters of VoIP may also be users of other phone equipment, such as PBX and cordless phone bases, that rely on power not provided by the telephone company. Even with local power still available, the broadband carrier itself may experience outages as well. While the PSTN has been matured over decades and is typically extremely reliable, most broadband networks are less than 10 years old, and even the best are still subject to intermittent outages. Furthermore, consumer network technologies such as cable and DSL often are not subject to the same restoration service levels as the PSTN or business technologies such as T-1 connection.

Quality of service

Some broadband connections may have less than desirable quality. Where IP packets are lost or delayed at any point in the network between VoIP users, there will be a momentary drop-out of voice. This is more noticeable in highly congested networks and/or where there are long distances and/or interworking between end points. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on.

It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, temporary failures have less impact on the communication quality. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable.

A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC3611 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 (due to jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, MOS scores and R factors and configuration information related to the jitter buffer.

RFC3611 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. RFC3611 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.

Difficulty with sending faxes

The support of sending faxes over VoIP is still limited. The existing voice codecs are not designed for fax transmission. An effort is underway to remedy this by defining an alternate IP-based solution for delivering Fax-over-IP, namely the T.38 protocol. Another possible solution to overcome the drawback is to treat the fax system as a message switching system which does not need 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.

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, and are impossible on some VoIP systems. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department. In the US, at least one major police department has strongly objected to this practice as potentially endangering the public.^[4]

Moreover, in the event that the caller is unable to give an address, emergency services may be unable to locate them in any other way. Following the lead of mobile phone operators, several VoIP carriers are already implementing a technical work-around.^{[citation needed]} For instance, one large VoIP carrier requires the registration of the physical address where the VoIP line will be used. When you dial the emergency number for your country, they will route it to the appropriate local system. They also maintain their own emergency call center that will take non-routable emergency calls (made, for example, from a software based service that is not tied to any particular physical location) and then will manually route your call once learning your physical location.^{[citation needed]}

e911 is another method by which VOIP providers in the US are able to support emergency services. The e911 emergency-calling system automatically associates a physical address with the calling party's telephone number as required by the Wireless Communications and Public Safety Act of 1999 and is being successfully used by many VOIP providers to provide physical address information to emergency service operators.

Integration into global telephone number system

While the traditional Plain Old Telephone Service (POTS) and mobile phone networks share a common global standard (E.164) which allocates and identifies any specific telephone line, there is no widely adopted similar standard for VoIP networks. Some allocate an E.164 number which can be used for VoIP as well as incoming/external calls. However, there are often different, incompatible schemes when calling between VoIP providers which use provider specific short codes.

Single point of calling

With hardware VoIP solutions it is possible to connect the VoIP router into the existing central phone box in the house and have VoIP at every phone already connected. Software based VoIP services require the use of a computer, so they are limited to single point of calling, though telephone sets are now available, allowing them to be used without a PC. Some services provide the ability to connect WiFi SIP phones so that service can be extended throughout the premises, and off-site to any location with an open hotspot.^[5] However, note that many hotspots require browser-based authentication, which most SIP phones do not support.^[6]

Mobile phones & Hand held Devices

Telcos and consumers have invested billions of dollars in mobile phone equipment. In developed countries, mobile phones have achieved nearly complete market penetration, and many people are giving up landlines and using mobiles exclusively. Given this situation, it is not entirely clear whether there would be a significant higher demand for VoIP among consumers until either public or community wireless networks have similar geographical coverage to cellular networks (thereby enabling mobile VoIP phones, so called WiFi phones or VoWLAN) or VoIP is implemented over 3G networks. However, "dual mode" telephone sets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular.^[7]

Phones like the NEC N900iL, and later many of the Nokia Eseries and several WiFi enabled mobile phones have SIP clients hardcoded into the firmware. Such clients operate independently of the mobile phone network unless a network operator decides to remove the client in the firmware of a heavily branded handset. Some operators such as Vodafone actively try to block VoIP traffic from their network^[8] and therefore most VoIP calls from such devices are done over WiFi.

Several WiFi only IP hardphones exist, most of them supporting either Skype or the SIP protocol. These phones are intended as a replacement for PSTN based cordless phones but can be used anywhere where WiFi internet access is available.

Another addition to hand held devices are ruggedized bar code type devices that are used in warehouses and retail environments. These type of devices rely on "inside the 4 walls" type of VoIP services that do not connect to the outside world and are solely to be used from employee to employee communications.

Security

Many consumer VoIP solutions do not support encryption yet, 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.^[9] There are several open source solutions that facilitate sniffing of VoIP conversations. A modicum of security is afforded due to patented audio codecs that are not easily available for open source applications, however such security through obscurity has not proven effective in the long run in other fields. Some vendors also use compression to make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely available at a consumer level. The existing secure standard SRTP and the new ZRTP protocol is 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.

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

Pre-Paid Phone Cards

VoIP has become an important technology for phone services to travelers, migrant workers and expatriates, who either, due to not having a fixed or mobile phone or high overseas roaming charges, choose instead to use VoIP services to make their phone calls. Pre-paid phone cards can be used either from a normal phone or from Internet cafes that have phone services. Developing countries and areas with high tourist or immigrant communities generally have a higher uptake.

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. When calling a traditional PSTN number from some VoIP providers, Caller ID is not supported.

In a few cases, VoIP providers may allow a caller to spoof the Caller ID information, making it appear as though they are calling from a different number. 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.

VoIM

Voice over Instant Messaging (VoIM) presents VoIP as one communication mode among several, with an IM user interface (contact list and presence) as the primary user experience. Many instant messenger services added client-to-client or client-to-PSTN VoIP in the mid-20

Voice over IP protocols 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 the caller directly to the VoIP user while access numbers require the caller to input the extension number of the VoIP user.