Voice over IP

VoIPvoice over Internet Protocolvoice-over-IPInternet telephonyvoiceIP telephonyVoice over IP (VoIP)digital phoneInternet callinginternet phone
Voice over Internet Protocol (VoIP), also called IP telephony, is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.wikipedia
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Internet

onlinethe Internetweb
Voice over Internet Protocol (VoIP), also called IP telephony, is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.
The Internet carries a vast range of information resources and services, such as the inter-linked hypertext documents and applications of the World Wide Web (WWW), electronic mail, telephony, and file sharing.

Telephony

digital telephonytelephonedigital
The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding.
In this context the technology is specifically referred to as Internet telephony, or voice over Internet Protocol (VoIP).

Unified communications

Unified Communicationcommunicationsunif
VoIP allows modern communications technologies (including telephones, smartphones, voice and video conferencing, email, and presence detection) to be consolidated using a single unified communications system.
Unified communications (UC) is a business and marketing concept describing the integration of enterprise communication services such as instant messaging (chat), presence information, voice (including IP telephony), mobility features (including extension mobility and single number reach), audio, web & video conferencing, fixed-mobile convergence (FMC), desktop sharing, data sharing (including web connected electronic interactive whiteboards), call control and speech recognition with non-real-time communication services such as unified messaging (integrated voicemail, e-mail, SMS and fax).

Session Initiation Protocol

SIPSession Initiation Protocol (SIP)SIP Trunking
Session Initiation Protocol (SIP), connection management protocol developed by the IETF
SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE).

Web-based VoIP

integrated into a web page
These protocols can be used by a VoIP phone, special-purpose software, a mobile application or integrated into a web page.
Web-based VoIP is the integration of voice over IP technologies into the facilities and methodologies of the World-Wide Web.

Speech coding

speech codecspeech encodingspeech
Various codecs exist that optimize the media stream based on application requirements and network bandwidth; some implementations rely on narrowband and compressed speech, while others support high-fidelity stereo codecs.
The two most important applications of speech coding are mobile telephony and voice over IP.

Federated VoIP

Third-generation providers, such as Google Talk, adopted the concept of federated VoIP—which is a departure from the architecture of the legacy networks.
Federated VoIP is a form of packetized voice telephony that uses voice over IP between autonomous domains in the public Internet without the deployment of central virtual exchange points or switching centers for traffic routing.

G.729

Some popular codecs include μ-law and A-law versions of G.711, G.722, an open source voice codec known as iLBC, a codec that uses only 8 kbit/s each way called G.729, and many others.
Because of its low bandwidth requirements, G.729 is mostly used in voice over Internet Protocol (VoIP) applications when bandwidth must be conserved, such as for conference calls.

XMPP

JabberExtensible Messaging and Presence ProtocolJabber/XMPP
Extensible Messaging and Presence Protocol (XMPP), instant messaging, presence information, and contact list maintenance
Designed to be extensible, the protocol has been used also for publish-subscribe systems, signalling for VoIP, video, file transfer, gaming, the Internet of Things (IoT) applications such as the smart grid, and social networking services.

Jingle (protocol)

JinglelibjingleGoogle Talk voice calls
Jingle, adds peer-to-peer session control to XMPP
Jingle is an extension to the Extensible Messaging and Presence Protocol (XMPP) which adds peer-to-peer (P2P) session control (signaling) for multimedia interactions such as in Voice over IP (VoIP) or videoconferencing communications.

H.323

RAS
H.323, one of the first VoIP call signaling and control protocols that found widespread implementation. Since the development of newer, less complex protocols such as MGCP and SIP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic.
A call model, similar to the ISDN call model, eases the introduction of IP telephony into existing networks of ISDN-based PBX systems, including transitions to IP-based PBXs.

Internet access

broadband internetbroadbandbroadband Internet access
Mass-market VoIP services use existing broadband Internet access, by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network (PSTN).
Telephony, radio, television, and videoconferencing

Media Gateway Control Protocol

MGCP
Media Gateway Control Protocol (MGCP), connection management for media gateways
The Media Gateway Control Protocol (MGCP) is a signaling and call control communications protocol used in voice over IP (VoIP) telecommunication systems.

Quality of service

QoSquality-of-serviceQuality of Service (QOS)
It is a best-effort network without fundamental Quality of Service (QoS) guarantees.
In particular, developers have introduced Voice over IP technology to allow computer networks to become as useful as telephone networks for audio conversations, as well as supporting new applications with even stricter network performance requirements.

Real-time Transport Protocol

RTPReal-Time ProtocolRTP / RTCP
Real-time Transport Protocol (RTP), transport protocol for real-time audio and video data
RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network.

Inter-Asterisk eXchange

IAXIAX2
Inter-Asterisk eXchange (IAX), protocol used between VoIP servers
It is used for transporting VoIP telephony sessions between servers and to terminal devices.

Skype protocol

Skype
Skype protocol, proprietary Internet telephony protocol suite based on peer-to-peer architecture
The Skype network is not interoperable with most other Voice over IP (VoIP) networks without proper licensing from Skype.

Skype protocol

Skype
Skype protocol, proprietary Internet telephony protocol suite based on peer-to-peer architecture
The Skype network is not interoperable with most other Voice over IP (VoIP) networks without proper licensing from Skype.

VoIP phone

IP PhoneIP PhonesEthernet-based phone
These protocols can be used by a VoIP phone, special-purpose software, a mobile application or integrated into a web page. In addition to VoIP phones, VoIP is also available on many personal computers and other Internet access devices.
A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet, instead of the traditional public switched telephone network (PSTN).

Session border controller

SBCSession Border Controllers
Private session border controllers are often employed to enable VoIP calls to and from protected networks.
A Session Border Controller (SBC) is a network element deployed to protect SIP based Voice over Internet Protocol (VoIP) networks.

Transmission Control Protocol

TCPTCP/IPACK
This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion.
Therefore, it is not particularly suitable for real-time applications such as voice over IP.

Caller ID

caller line identificationCLIcall display
Voice over IP protocols and equipment provide caller ID support that is compatible with the facility provided in the public switched telephone network (PSTN).
Caller ID (caller identification, CID), also called calling line identification (CLID), Calling Line Identification (CLI), calling number delivery (CND), calling number identification (CNID), calling line identification presentation (CLIP), or call display, is a telephone service, available in analog and digital telephone systems, including VoIP, that transmits a caller's telephone number to the called party's telephone equipment when the call is being set up.

Analog telephone adapter

analog telephony adapterATAAnalogue Telephone Adaptors
An analog telephone adapter connects to the network and implements the electronics and firmware to operate a conventional analog telephone attached through a modular phone jack. Some residential Internet gateways and cablemodems have this function built in.
An analog telephone adapter (ATA) is a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephony network.

Internet telephony service provider

ITSPsNet telephonyVoIP services
In such cases, the Internet telephony service provider (ITSP) knows only that a particular user's equipment is active.
An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet.

VoIP vulnerabilities

This means that hackers with knowledge of VoIP vulnerabilities can perform denial-of-service attacks, harvest customer data, record conversations, and compromise voicemail messages.
VoIP is vulnerable to similar types of attacks that Web connection and emails are prone to. VoIP attractiveness, because of its low fixed cost and numerous features, come with some risks that are well known to the developers an are constantly being addressed.