Voice Over Internet Protocol ( VOIP )

Voice over Internet Protocol (VoIP, IPA: /v??p/) is a general term for a family of transmission technologies for delivery of voice communications over the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VOIP phone systems are IP telephony and Internet telephony, as well as voice over broadband, broadband telephony, and broadband phone, when the network connectivity is available over broadband Internet access.

VoIP systems usually interface with the traditional public switched telephone network (PSTN) to allow for transparent phone communications worldwide[1].

VoIP can be a benefit for reducing communication and infrastructure costs by routing phone calls over existing data networks and avoiding duplicate network systems.

Voice-over-IP systems carry telephony speech as digital audio, typically reduced in data rate using speech data compression techniques, packetized in small units of typically tens of milliseconds of speech, and encapsulated in a packet stream over IP.

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.[2] 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 possible to speak only to other Vocaltec Internet Phone users.[3] 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.[4]

Revenue in the total 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.


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, computer 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.

Implementation

Because the underlying IP network is inherently unreliable, in contrast to the circuit-switched public telephone network, and does not inherently provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (QoS) guarantees, VoIP implementations face problems mitigating 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
Some analog telephone adapters do not decode pulse dialing from older phones. The VoIP user may use a 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 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 analog audio, although this further increases delay. This avoids a condition known as buffer underrun, in which the voice engine is missing audio since the next voice packet has not yet arrived.

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 backup generators or batteries located at the telephone exchange. However, IP Phones and the IP infrastructure connect to (routers and servers), which typically depend on the availability of mains electricity or another locally generated power source. Therefore, most VoIP networks and the supporting routers and servers are also on widely available and relatively inexpensive uninterrupted power supply (UPS) systems to maintain electricity during a power outage for a predetermined length of time. The amount of time typically ranges from as little as an hour and up from there, depending on the quality of the UPS unit and the power draw and characteristics of the communications equipment.

Voice travels over the Internet in packets in the same manner as data. So when you talk over an IP network your conversation is broken up into small packets. These voice and data packets travel over the same network with a fixed bandwidth. This system is more prone to congestion and DoS attacks[5] than traditional circuit switched systems.[citation needed]


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 between end points. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance.

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.