Voice over IP

Voice over Internet Protocol (Voice over IP, VoIP and 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 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).

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. Instead of being transmitted over a circuit-switched network; however, the digital information is packetized, and transmission occurs as IP packets over a packet-switched network. They transport audio streams using special media delivery protocols that encode audio and video with audio codecs, and video codecs. 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. Some popular codecs include μ-law and a-law versions of G.711, G.722, a popular open source voice codec known as iLBC, a codec that only uses 8 kbit/s each way called G.729, and many others.

Early providers of voice-over-IP services offered business models and technical solutions that mirrored the architecture of the legacy telephone network. Second-generation providers, such as Skype, have 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. This has limited the freedom of users to mix-and-match third-party hardware and software. Third-generation providers, such as Google Talk, have adopted the concept of federated VoIP—which is a departure from the architecture of the legacy networks. These solutions typically allow dynamic interconnection between users on any two domains on the Internet when a user wishes to place a call.

In addition to VoIP phones, VoIP is available on many smartphones, personal computers, and on Internet access devices. Calls and SMS text messages may be sent over 3G/4G or Wi-Fi.

Protocols

Voice over IP has been implemented in various ways using both proprietary protocols and protocols based on open standards. VoIP protocols include:

• Session Initiation Protocol (SIP)

• H.323

• Media Gateway Control Protocol (MGCP)

• Gateway Control Protocol (Megaco, H.248)

• Real-time Transport Protocol (RTP)

• Real-time Transport Control Protocol (RTCP)

• Secure Real-time Transport Protocol (SRTP)

• Session Description Protocol (SDP)

• Inter-Asterisk eXchange (IAX)

• Jingle XMPP VoIP extensions

• Skype protocol

The H.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic, as well as local area network services. However, 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.

These protocols can be used by special-purpose software, such as Jitsi, or integrated into a web page or a mobile application (web-based VoIP), like Google Talk.