VoIP, short for Voice over Internet Protocol, is a set of technologies that send voice calls as data packets over Internet Protocol (IP) networks, including the Internet. A regular telephone can join a VoIP service through an analog telephone adapter (ATA). VoIP is also called IP telephony.
History of VoIP
Creator: Several developers and standards bodies, including the ITU-T and the IETF. Year: 1974.
Voice over IP (VoIP) work started on the early ARPANET. In 1974 the first real-time conversation on ARPANET used 2.4 kbps (kilobits per second). It used linear predictive coding (LPC). It ran between two sites. One was Culler-Harrison Incorporated in Goleta, California. The other was MIT Lincoln Laboratory in Lexington, Massachusetts. In 1977, Danny Cohen, Jon Postel and Vint Cerf agreed to separate IP from TCP. IP means Internet Protocol. TCP means Transmission Control Protocol. They also agreed to create UDP for real-time traffic. UDP means User Datagram Protocol.
Speak Freely, a VoIP app, was released to the public domain in 1991. VocalTec released Internet Phone in 1995. The International Telecommunication Union (ITU-T) began work in 1996. That work was on the H.323 standard. The Session Initiation Protocol (SIP) specification is RFC 2543. It was released in 1999. The Internet Engineering Task Force (IETF) published RFC 3261 for SIP in June 2002.
In 1999 Mark Spencer of Digium developed Asterisk. Asterisk is open source software for a private branch exchange (PBX). Skype was first released on 29 August 2003. Niklas Zennström and Janus Friis created it with four Estonian developers.
Mobile networks added VoIP voice as Voice over LTE (VoLTE) with 4G. Voice over 5G (Vo5G) came with 5G. WebRTC rose in 2011 and brought VoIP into web browsers. The Opus codec was introduced in 2012. The Wikipedia timeline lists SILK in 2009. It lists the AAC-LD codec, used by FaceTime, in 2010.
Changelog and latest update
VoIP has no single version number. Its standards change through RFCs. RFC means Request for Comments, a numbered document series. The Internet Engineering Task Force (IETF) publishes them. The SIP standard, RFC 3261, was published in June 2002. The RFC Editor page for that standard lists 21 later RFCs that update it.
H.323 was one of the first VoIP call signaling protocols in wide use. Simpler protocols, such as the Media Gateway Control Protocol (MGCP) and SIP, replaced it in new systems. H.323 now carries mostly existing long-haul traffic. The Wikipedia article says so.
Secure Real-time Transport Protocol (SRTP) is the encrypted form of the Real-time Transport Protocol (RTP). RFC 3711 defined SRTP in March 2004. The Session Description Protocol (SDP) was published in RFC 4566 in July 2006. RFC 8866 later replaced it. These documents define how VoIP sets up, encrypts and sends each call.
Latest update: RFC 8996 (TLS 1.0 and TLS 1.1 deprecation), 2021-03
RFC 8996 was published in March 2021. It deprecates TLS 1.0 and TLS 1.1. Its header lists RFC 3261 among the documents it updates. The RFC Editor page for RFC 3261 lists 21 later RFCs that update it. RFC 8996 is the newest of those 21 by date. RFC 8898 from September 2020 is another update to RFC 3261.
Usage examples
The examples show the text that VoIP systems send and read. SIP messages set up a call, SDP messages describe the audio codecs, Asterisk dialplan lines route the call, and an rtp.conf setting sets the media port range. The SIP and SDP formats follow RFC 3261 and RFC 4566. Each example runs on a server or a phone system that supports the protocol.
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/UDP pc33.example.com
;branch=z9hG4bK776
Max-Forwards: 70
To: <sip:bob@example.com>
From: <sip:alice@example.com>;tag=1928
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Content-Length: 0This SIP INVITE request asks Bob to start a call from Alice. The Via header holds the branch value that identifies this transaction. The Call-ID and CSeq headers number the request, so the server can match each reply to it. The Content-Length of zero means this request carries no session body.
v=0
o=alice 2890844526 1 IN IP4 192.0.2.10
s=Call
c=IN IP4 192.0.2.10
t=0 0
m=audio 49170 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000This Session Description Protocol (SDP) block describes an audio stream sent to address 192.0.2.10 on port 49170. Payload type 0 is PCMU and type 8 is PCMA, which are the mu-law and A-law forms of the G.711 codec. The receiving side picks one codec from the list.
[default]
exten => 100,1,Dial(SIP/alice)
exten => 100,n,Hangup()The Asterisk dialplan routes a call to extension 100. The Dial application sends the call to the SIP endpoint named alice. The Hangup line ends the call when the Dial step finishes. Asterisk is open source PBX software that many small phone systems use.
[general]
rtpstart=10000
rtpend=20000The rtp.conf file sets the range of UDP ports that Asterisk uses for the Real-time Transport Protocol (RTP) audio stream. Each call takes two ports from this range. The firewall must allow this range for UDP traffic, or the call has no audio.
References
These sources are outside Websnoogie and were checked on 2026-10-12.
Voice over IP
Wikipedia
The encyclopedia article on VoIP. It covers the definition, the codecs, the protocols such as SIP, H.323 and RTP, the business models, and a timeline from the 1974 ARPANET conversation to the 2012 Opus codec, with citations to the sources.
RFC 3261: SIP, Session Initiation Protocol
RFC Editor (IETF)
The IETF standard for SIP, published in June 2002. It defines how a session is created, changed and ended, and how users register their location. The page lists the RFCs that update it, with 21 entries.
RFC 3550: RTP, Real-time Transport Protocol
RFC Editor (IETF)
The IETF standard for RTP, the protocol that carries the audio and video of a VoIP call. It also defines RTCP, the control protocol that reports the statistics and status of each media stream to the endpoints.
RFC 3711: SRTP
RFC Editor (IETF)
The IETF standard for the Secure Real-time Transport Protocol, published in March 2004. It defines encryption and message authentication for RTP packets. The page lists RFC 9335, RFC 5506 and RFC 6904 as later updates.
RFC 4566: SDP, Session Description Protocol
RFC Editor (IETF)
The IETF standard for SDP, published in July 2006. It defines the text format that describes the media, addresses and ports of a session. The page notes that RFC 8866 obsoletes it, so the current version is RFC 8866.
Phone lines or VoIP calls: how to compare
A traditional phone line runs on copper wire from the phone company. VoIP, which stands for voice over internet protocol, sends the same call over your internet connection. The two cost different amounts and fail in different ways. A copper line can keep working when the internet is down. A VoIP line depends on the connection and the router. Small businesses should compare both before they print a new number on their website.
For your website, the number matters more than the technology behind it. We place the number in the header, on the contact page and in the structured data, so search engines and visitors find it. On phones, we make the number tap-to-call, which saves a visitor from copying digits by hand. We have worked on business websites for more than 20 years, and a hidden number sends a visitor elsewhere.
Phone 402-813-4034 to talk through your phone setup. Or order a new website for $300, one time, with five pages and a contact form, delivered in 3 to 5 days. We will tell you where the number should appear first.
