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).

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.

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).

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.

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.

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.

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.

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.

Internet Low Bitrate Codec

iLBCGIPS codec
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.
It is suitable for VoIP applications, streaming audio, archival and messaging.

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.

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

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.

Fax

fax machinefacsimilefax machines
The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).
T.38 is designed to work with VoIP services and often supported by analog telephone adapters used by legacy fax machines that need to connect through a VoIP service.

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.

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.

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.

Softphone

soft phonePC-to-PC calling
Softphone application software installed on a networked computer that is equipped with a microphone and speaker, or headset. The application typically presents a dial pad and display field to the user to operate the application by mouse clicks or keyboard input.
To communicate, both end-points must support the same voice-over-IP protocol, and at least one common audio codec.

Skype

SkypingSkype PreviewMicrosoft Skype
Second-generation providers, such as Skype, built closed networks for private user bases, offering the benefit of free calls and convenience while potentially charging for access to other communication networks, such as the PSTN.
Skype uses a proprietary Internet telephony (VoIP) network called the Skype protocol.

G.722

ITU-T G.722
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.
This is useful for voice over IP applications, such as on a local area network where network bandwidth is readily available, and offers a significant improvement in speech quality over older narrowband codecs such as G.711, without an excessive increase in implementation complexity.

Google Talk

Google ChatGTalkchat
Third-generation providers, such as Google Talk, adopted the concept of federated VoIP—which is a departure from the architecture of the legacy networks.
On December 15, 2005, Google released libjingle, a C++ library to implement Jingle, "a set of extensions to the IETF's Extensible Messaging and Presence Protocol (XMPP) for use in voice over IP (VoIP), video, and other peer-to-peer multimedia sessions."

Smartphone

smartphonessmart phonesmart phones
Smartphones may have SIP clients built into the firmware or available as an application download.
Audio quality can be improved using a VoIP application over WiFi.