Editing Voice over IP (section)

==History==
The early developments of [[packet network]] designs by [[Paul Baran]] and other researchers were motivated by a desire for a higher degree of circuit redundancy and network availability in the face of infrastructure failures than was possible in the circuit-switched networks in [[telecommunications]] of the mid-twentieth century. [[Danny Cohen (engineer)|Danny Cohen]] first demonstrated a form of [[packet telephony|packet voice]] in 1973 which was developed into [[Network Voice Protocol]] which operated across the early [[ARPANET]].<ref>{{cite web|url=http://www.internethalloffame.org/inductees/danny-cohen|title=Danny Cohen |publisher=INTERNET HALL of FAME|access-date=December 6, 2014}}</ref><ref>{{cite book|url=https://books.google.com/books?id=3yaUBAAAQBAJ&q=Network+Voice+Protocol+%28NVP%29+developed+by+Danny+Cohen&pg=PA34|publisher=Willey|title=Advanced Content Delivery, Streaming, and Cloud Services (Pg 34)|access-date=December 6, 2014|isbn=9781118909706|date=September 19, 2014}}</ref>

On the early ARPANET, real-time voice communication was not possible with uncompressed [[pulse-code modulation]] (PCM) [[digital audio|digital speech]] packets, which had a [[bit rate]] of 64{{nbsp}}kbps, much greater than the 2.4{{nbsp}}kbps [[Internet bandwidth|bandwidth]] of early [[modems]]. The solution to this problem was [[linear predictive coding]] (LPC), a [[speech coding]] [[data compression]] algorithm that was first proposed by [[Fumitada Itakura]] of [[Nagoya University]] and Shuzo Saito of [[Nippon Telegraph and Telephone]] (NTT) in 1966. LPC was capable of speech compression down to 2.4{{nbsp}}kbps, leading to the first successful real-time conversation over ARPANET in 1974, between Culler-Harrison Incorporated in [[Goleta, California]], and [[MIT Lincoln Laboratory]] in [[Lexington, Massachusetts]].<ref name="Gray">{{cite journal |last1=Gray |first1=Robert M. |title=A History of Realtime Digital Speech on Packet Networks: Part II of Linear Predictive Coding and the Internet Protocol |journal=Found. Trends Signal Process. |date=2010 |volume=3 |issue=4 |pages=203–303 |doi=10.1561/2000000036 |url=https://ee.stanford.edu/~gray/lpcip.pdf |issn=1932-8346|doi-access=free }}</ref> LPC has since been the most widely used speech coding method.<ref>{{cite journal |last1=Gupta |first1=Shipra |title=Application of MFCC in Text Independent Speaker Recognition |journal=International Journal of Advanced Research in Computer Science and Software Engineering |date=May 2016 |volume=6 |issue=5 |pages=805–810 (806) |s2cid=212485331 |issn=2277-128X |url=https://pdfs.semanticscholar.org/2aa9/c2971342e8b0b1a0714938f39c406f258477.pdf |archive-url=https://web.archive.org/web/20191018231621/https://pdfs.semanticscholar.org/2aa9/c2971342e8b0b1a0714938f39c406f258477.pdf |url-status=dead |archive-date=October 18, 2019 |access-date=October 18, 2019}}</ref> [[Code-excited linear prediction]] (CELP), a type of LPC algorithm, was developed by [[Manfred R. Schroeder]] and [[Bishnu S. Atal]] in 1985.<ref name="Schroeder">M. R. Schroeder and B. S. Atal, "Code-excited linear prediction (CELP): high-quality speech at very low bit rates," in ''Proceedings of the IEEE [[International Conference on Acoustics, Speech, and Signal Processing]]'' (ICASSP), vol. 10, pp.&nbsp;937–940, 1985.</ref> LPC algorithms remain an [[audio coding standard]] in modern VoIP technology.<ref name="Gray"/>

In the two decades following the 1974 demo, various forms of packet telephony were developed and industry interest groups formed to support the new technologies. Following the termination of the ARPANET project, and expansion of the [[Internet]] for commercial traffic, IP telephony was tested and deemed infeasible for commercial use until the introduction of VocalChat in the early 1990s and then in Feb 1995 the official release of Internet Phone (or iPhone for short) commercial software by [[VocalTec]], based on a patent by [[Lior Haramaty]] and [[Alon Cohen]],<ref>[https://patents.google.com/patent/US5825771 Audio Transceiver]</ref> and followed by other VoIP infrastructure components such as telephony gateways and switching servers. Soon after it became an established area of interest in commercial labs of the major IT concerns, notably at AT&T, where [[Marian Croak]] and her team filed many patents related to the technology.{{cn|reason=[[Marian Croak]] and sources there are not clear on exactly when this work occurred.|date=March 2024}} By the late 1990s, the first [[softswitch]]es became available, and new protocols, such as [[H.323]], MGCP and [[Session Initiation Protocol]] (SIP) gained widespread attention. In the early 2000s, the proliferation of high-bandwidth always-on Internet connections to residential dwellings and businesses, spawned an industry of Internet telephony service providers (ITSPs). The development of open-source telephony software, such as [[Asterisk PBX]], fueled widespread interest and entrepreneurship in voice-over-IP services, applying new Internet technology paradigms, such as [[cloud service]]s to telephony.

===Milestones===
* 1966: [[Linear predictive coding]] (LPC) proposed by [[Fumitada Itakura]] of [[Nagoya University]] and Shuzo Saito of [[Nippon Telegraph and Telephone]] (NTT).<ref name="Gray"/>
* 1973: [[Packet telephony|Packet voice]] application by [[Danny Cohen (engineer)|Danny Cohen]].
* 1974: The [[Institute of Electrical and Electronics Engineers]] (IEEE) publishes a paper entitled "A Protocol for Packet Network Interconnection".<ref>{{cite journal|last=Cerf|first=V.|author2=Kahn, R. |title=A Protocol for Packet Network Intercommunication|journal=IEEE Transactions on Communications|date=May 1974|volume=22|issue=5|pages=637–648|doi=10.1109/TCOM.1974.1092259|url=http://www.cs.rice.edu/~eugeneng/teaching/f07/comp529/papers/ck74.pdf}}</ref>
* 1974: [[Network Voice Protocol]] (NVP) tested over [[ARPANET]] in August 1974, carrying barely intelligible 16{{nbsp}}kpbs [[CVSD]] encoded voice.<ref name="Gray"/>
* 1974: The first successful real-time conversation over ARPANET achieved using 2.4{{nbsp}}kpbs LPC, between Culler-Harrison Incorporated in [[Goleta, California]], and [[MIT Lincoln Laboratory]] in [[Lexington, Massachusetts]].<ref name="Gray"/>
* 1977: Danny Cohen and [[Jon Postel]] of the USC [[Information Sciences Institute]], and [[Vint Cerf]] of the Defense Advanced Research Projects Agency (DARPA), agree to separate IP from TCP, and create UDP for carrying real-time traffic.
* 1981: [[IPv4]] is described in RFC 791.
* 1985: The [[National Science Foundation]] commissions the creation of [[NSFNET]].<ref>{{cite web|url=https://www.nsf.gov/about/history/nsf0050/internet/launch.htm|title=The Launch of NSFNET|publisher=The National Science Foundation|access-date=January 21, 2009|archive-date=May 7, 2006|archive-url=https://web.archive.org/web/20060507225813/http://www.nsf.gov/about/history/nsf0050/internet/launch.htm|url-status=dead}}</ref>
* 1985: [[Code-excited linear prediction]] (CELP), a type of LPC algorithm, developed by [[Manfred R. Schroeder]] and [[Bishnu S. Atal]].<ref name="Schroeder"/>
* 1986: Proposals from various standards organizations{{specify|date=April 2012}} for [[VoATM|Voice over ATM]], in addition to commercial packet voice products from companies such as [[StrataCom]]
* 1991: Speak Freely, a voice-over-IP application, was released to the public domain.<ref name="B2C">{{cite web |last1=Dua |first1=Amit |title=VoIP Basics: Everything Beginners Should Know! |url=https://www.business2community.com/communications/voip-basics-everything-beginners-should-know-02422214 |website=business2community.com |date=July 29, 2021 |publisher=Business 2 Community |access-date=14 September 2021}}</ref><ref name="McCraw">{{cite web |last1=McCraw |first1=Corey |title=The History of VoIP Over the Last 55 Years (1966 to 2021) |url=https://fitsmallbusiness.com/history-of-voip/ |website=fitsmallbusiness.com |date=October 12, 2022 |publisher=Fits Small Business}}</ref>
* 1992: The Frame Relay Forum conducts development of standards for voice over [[Frame Relay]].
* 1992: [[InSoft Inc.]] announces and launches its desktop conferencing product Communique, which includes VoIP and video.<ref name="B2C" /><ref name="Inc1992">{{cite journal|author1=IDG Network World Inc|last2=Eckerson|first2=Wayne| title=Network World - Startup targets desktop Videoconferencing arena|journal=Network World|url=https://books.google.com/books?id=DhQEAAAAMBAJ&pg=PA39|access-date=February 10, 2012|date=September 21, 1992|publisher=IDG Network World Inc|pages=39–|issn=0887-7661}}</ref> The company is credited with developing the first generation of commercial, US-based VoIP, Internet media streaming and real-time Internet telephony/collaborative software and standards that would provide the basis for the Real Time Streaming Protocol (RTSP) standard.{{cn|date=July 2023}}
* 1993 Release of VocalChat, a commercial packet network PC voice communication software from [[VocalTec]].{{cn|reason=no occurrence of VocalChat at [[VocalTec]]|date=July 2023}}
* 1994: MTALK, a freeware LAN VoIP application for [[Linux]]<ref>{{cite web |url=http://sunsite.unc.edu/pub/Linux/apps/sound/talk/mtalk.README |publisher=Sunsite.edu |title=MTALK-Readme |access-date=April 29, 2012 |format=TXT}}</ref>
* 1995: 
**[[VocalTec]] releases ''Internet Phone'' commercial Internet phone software.<ref>{{cite web |url=http://blog.tmcnet.com/blog/tom-keating/docs/cti-buyers-guide-1996.pdf |publisher=Computer Telephony Interaction Magazine |title=Internet Phone Release 4 |last=Keating |first=Tom |access-date=November 7, 2007}}</ref><ref>{{cite web |url=http://www.ilocus.com/2007/07/the_10_that_established_voip_p_2.html |publisher=iLocus |title=The 10 that Established VOIP (Part 1: VocalTec) |access-date=January 21, 2009}}</ref>
** [[Intel]], [[Microsoft]] and [[Radvision]] initiated standardization activities for VoIP communications system.<ref>The free Library [http://www.thefreelibrary.com/RADVision+and+Intel+Target+Compatibility+Between+RADVision%27s+...-a019467970 RADVision and Intel Target Compatibility Between RADVision's H.323/320 Videoconferencing Gateway And Intel's Business Video Conferencing And TeamStation Products.] {{Webarchive|url=https://web.archive.org/web/20131030095918/http://www.thefreelibrary.com/RADVision+and+Intel+Target+Compatibility+Between+RADVision%27s+...-a019467970 |date=October 30, 2013 }} June 2, 1997 [http://www.radvision.com/Developer-Solutions/ VoiP Developer Solutions] {{Webarchive|url=https://web.archive.org/web/20110616001931/http://www.radvision.com/Developer-Solutions |date=June 16, 2011 }}</ref>
* 1996:
** [[ITU-T]] begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the [[H.323]] standard.<ref>{{cite web |url=http://www.itu.int/rec/T-REC-H.323-199611-S/en |title=H.323 Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service |publisher=ITU-T |access-date=January 21, 2009}}</ref>
** US telecommunications companies petition the US Congress to ban Internet phone technology.<ref>{{cite web|url=http://www.faqs.org/rfcs/rfc2235.html|title=RFC 2235|publisher=R. Zakon|access-date=January 21, 2009}}</ref>
** [[G.729]] speech codec introduced, using CELP (LPC) algorithm.<ref>International Telecommunication Union, Standardization Sector (ITU-T), Study Group 15 (1993-1996), ''Recommendation G.729'', March 1996.</ref>
* 1997: [[Level 3 Communications|Level 3]] began development of its first [[softswitch]], a term they coined in 1998.<ref name="ilocus softswitch">{{cite web|url=http://www.ilocus.com/2007/07/the_10_that_established_voip_p_1.html|title=The 10 that Established VOIP (Part 2: Level 3)|publisher=iLocus|date=July 13, 2007|access-date=November 7, 2007}}</ref>
* 1999:
** The [[Session Initiation Protocol]] (SIP) specification RFC 2543 is released.<ref>{{cite web|url=http://www.ietf.org/rfc/rfc2543.txt|title=RFC 2543, SIP: Session Initiation Protocol|publisher=Handley, Schulzrinne, Schooler, Rosenberg|access-date=January 21, 2009}}</ref>
** [[Mark Spencer (computer engineer)|Mark Spencer]] of [[Digium]] develops [[Asterisk (PBX)|Asterisk]], the first [[Open source software|open source]] [[private branch exchange]] (PBX) software.<ref>{{cite web|url=http://www.asterisk.org/about|title=What is Asterisk|publisher=Asterisk.org|access-date=January 21, 2009|archive-date=January 23, 2009|archive-url=https://web.archive.org/web/20090123154417/http://www.asterisk.org/about|url-status=dead}}</ref>
** A [[discrete cosine transform]] (DCT) variant called the [[modified discrete cosine transform]] (MDCT) is adopted for the [[Siren (codec)|Siren]] codec, used in the [[G.722.1]] [[wideband audio]] coding standard.<ref name="Hersent">{{cite book |last1=Hersent |first1=Olivier |last2=Petit |first2=Jean-Pierre |last3=Gurle |first3=David |title=Beyond VoIP Protocols: Understanding Voice Technology and Networking Techniques for IP Telephony |date=2005 |publisher=[[John Wiley & Sons]] |isbn=9780470023631 |page=55 |url=https://books.google.com/books?id=SMvNToRs-DgC&pg=PA55}}</ref><ref name="Lutzky">{{cite conference |last1=Lutzky |first1=Manfred |last2=Schuller |first2=Gerald |last3=Gayer |first3=Marc |last4=Krämer |first4=Ulrich |last5=Wabnik |first5=Stefan |title=A guideline to audio codec delay |url=https://www.iis.fraunhofer.de/content/dam/iis/de/doc/ame/conference/AES-116-Convention_guideline-to-audio-codec-delay_AES116.pdf |website=[[Fraunhofer IIS]] |conference=116th AES Convention |publisher=[[Audio Engineering Society]] |date=May 2004 |access-date=October 24, 2019}}</ref>
** The MDCT is adapted into the LD-MDCT algorithm, used in the [[AAC-LD]] standard.<ref name="Schnell">{{cite conference |last1=Schnell |first1=Markus |last2=Schmidt |first2=Markus |last3=Jander |first3=Manuel |last4=Albert |first4=Tobias |last5=Geiger |first5=Ralf |last6=Ruoppila |first6=Vesa |last7=Ekstrand|first7=Per |last8=Bernhard |first8=Grill |date=October 2008 |title=MPEG-4 Enhanced Low Delay AAC - A New Standard for High Quality Communication |url=https://www.iis.fraunhofer.de/content/dam/iis/de/doc/ame/conference/AES-125-Convention_AAC-ELD-NewStandardForHighQualityCommunication_AES7503.pdf |conference=125th AES Convention |publisher=[[Audio Engineering Society]] |access-date=October 20, 2019 |website=[[Fraunhofer IIS]]}}</ref>
* 2001: [[INOC-DBA]], the first inter-provider [[Session Initiation Protocol|SIP]] network is deployed; this is also the first voice network to reach all seven continents.<ref>{{cite book |last1=Stapleton-Gray |first1=Ross |title=Inter-Network Operations Center Dial-by-ASN (INOC-DBA), a Resource for the Network Operator Community |date=2009 |publisher=IEEE Computer Society Press |location=Los Alamitos |isbn=978-0-7695-3568-5}}</ref>
* 2003: [[Skype]] released in August 2003. This was the creation of Niklas Zennström and Janus Friis, in cooperation with four Estonian developers. It quickly became a popular program that helped democratize VoIP. 
* 2004: Early commercial VoIP service providers proliferate.{{cn|date=December 2024}}
* 2005: [[PhoneGnome]]  VoIP service is launched by TelEvolution, Inc. of California.<ref>{{cite news|url=https://www.nytimes.com/2007/08/02/technology/circuits/02pogue.html|title=State of the Art: Get Your Free Net Phone Calls Here|last=Pogue|first=David|date=August 2, 2007|work=The New York Times|accessdate=2009-01-20}}</ref>
* 2006: [[G.729.1]] wideband codec introduced, using MDCT and CELP (LPC) algorithms.<ref name="Nagireddi">{{cite book |last1=Nagireddi |first1=Sivannarayana |title=VoIP Voice and Fax Signal Processing |date=2008 |publisher=[[John Wiley & Sons]] |isbn=9780470377864 |page=69 |url=https://books.google.com/books?id=5AneeZFE71MC&pg=PA69}}</ref>
* 2007: VoIP device manufacturers and sellers boom in Asia, specifically in the Philippines where many families of overseas workers reside.<ref>{{Cite news|url=https://news.google.com/newspapers?nid=2479&dat=20070827&id=j1M1AAAAIBAJ&pg=1974,4860651|title=Prospects bright for voice calls over internet|last=Remo|first=Michelle V.|date=August 27, 2007|newspaper=Philippine Daily Inquirer|access-date=January 1, 2015}}</ref>
* 2009: [[SILK]] codec introduced, using LPC algorithm,<ref name="IETF79">[http://nagasaki.bogus.com/ietf79/ietf79-ch8-tue-noon.mp3 Audio-Mitschnitt] {{webarchive|url=https://web.archive.org/web/20130210051956/http://nagasaki.bogus.com/ietf79/ietf79-ch8-tue-noon.mp3 |date=February 10, 2013 }} vom Treffen der IETF-Codec-Arbeitsgruppe auf der Konferenz IETF79 in Peking, China mit einer Darstellung der grundlegenden Funktionsprinzipien durch Koen Vos (MP3, ~70&nbsp;MiB)</ref> and used for voice calling in [[Skype]].<ref>{{cite web |url=http://www.wirevolution.com/2009/01/13/skypes-new-super-wideband-codec/ |title=Skype's new super wideband codec |publisher=Wirevolution.com |date=January 13, 2009 |access-date=March 31, 2009}}</ref>
* 2010: [[Apple Inc.|Apple]] introduces [[FaceTime]], which uses the LD-MDCT-based AAC-LD codec.<ref name="AppleInsider standards 1">{{cite web|url=http://www.appleinsider.com/articles/10/06/08/inside_iphone_4_facetime_video_calling.html|date=June 8, 2010|access-date=June 9, 2010|title=Inside iPhone 4: FaceTime video calling|publisher=[[Apple community#AppleInsider|AppleInsider]]|author=Daniel Eran Dilger}}</ref>
* 2011:
** Rise of [[WebRTC]] technology which supports VoIP directly in browsers.
** [[CELT]] codec introduced, using MDCT algorithm.<ref name="presentation">[http://people.xiph.org/~greg/video/linux_conf_au_CELT_2.ogv Presentation of the CELT codec] {{Webarchive|url=https://web.archive.org/web/20110807182250/http://people.xiph.org/~greg/video/linux_conf_au_CELT_2.ogv |date=August 7, 2011 }} by Timothy B. Terriberry (65 minutes of video, see also [http://www.celt-codec.org/presentations/misc/lca-celt.pdf presentation slides] {{Webarchive|url=https://web.archive.org/web/20110810032741/http://www.celt-codec.org/presentations/misc/lca-celt.pdf |date=August 10, 2011 }} in PDF)</ref>
* 2012: [[Opus (audio format)|Opus]] codec introduced, using MDCT and LPC algorithms.<ref name="Valin">{{cite conference|last1=Valin|first1=Jean-Marc|last2=Maxwell|first2=Gregory|last3=Terriberry|first3=Timothy B.|last4=Vos|first4=Koen|date=October 2013|title=High-Quality, Low-Delay Music Coding in the Opus Codec|conference=135th AES Convention|publisher=[[Audio Engineering Society]]|arxiv=1602.04845}}</ref>