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Voice over IP - Wikipedia
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cdx-button--weight-quiet cdx-button--icon-only vector-toc-toggle"> <span class="vector-icon mw-ui-icon-wikimedia-expand"></span> <span>Toggle Adoption subsection</span> </button> <ul id="toc-Adoption-sublist" class="vector-toc-list"> <li id="toc-Consumer_market" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Consumer_market"> <div class="vector-toc-text"> <span class="vector-toc-numb">3.1</span> <span>Consumer market</span> </div> </a> <ul id="toc-Consumer_market-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-PSTN_and_mobile_network_providers" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#PSTN_and_mobile_network_providers"> <div class="vector-toc-text"> <span class="vector-toc-numb">3.2</span> <span>PSTN and mobile network providers</span> </div> </a> <ul id="toc-PSTN_and_mobile_network_providers-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Corporate_use" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Corporate_use"> <div class="vector-toc-text"> <span class="vector-toc-numb">3.3</span> <span>Corporate use</span> </div> </a> <ul id="toc-Corporate_use-sublist" class="vector-toc-list"> </ul> </li> </ul> </li> <li id="toc-Delivery_mechanisms" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Delivery_mechanisms"> <div class="vector-toc-text"> <span class="vector-toc-numb">4</span> <span>Delivery mechanisms</span> </div> </a> <button aria-controls="toc-Delivery_mechanisms-sublist" class="cdx-button cdx-button--weight-quiet cdx-button--icon-only vector-toc-toggle"> <span class="vector-icon mw-ui-icon-wikimedia-expand"></span> <span>Toggle Delivery mechanisms subsection</span> </button> <ul id="toc-Delivery_mechanisms-sublist" class="vector-toc-list"> <li id="toc-Hosted_VoIP_systems" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Hosted_VoIP_systems"> <div class="vector-toc-text"> <span class="vector-toc-numb">4.1</span> <span>Hosted VoIP systems</span> </div> </a> <ul id="toc-Hosted_VoIP_systems-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Private_VoIP_systems" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Private_VoIP_systems"> <div class="vector-toc-text"> <span class="vector-toc-numb">4.2</span> <span>Private VoIP systems</span> </div> </a> <ul id="toc-Private_VoIP_systems-sublist" class="vector-toc-list"> </ul> </li> </ul> </li> <li id="toc-Quality_of_service" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Quality_of_service"> <div class="vector-toc-text"> <span class="vector-toc-numb">5</span> <span>Quality of service</span> </div> </a> <button aria-controls="toc-Quality_of_service-sublist" class="cdx-button cdx-button--weight-quiet cdx-button--icon-only vector-toc-toggle"> <span class="vector-icon mw-ui-icon-wikimedia-expand"></span> <span>Toggle Quality of service subsection</span> </button> <ul id="toc-Quality_of_service-sublist" class="vector-toc-list"> <li id="toc-DSL_and_ATM" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#DSL_and_ATM"> <div class="vector-toc-text"> <span class="vector-toc-numb">5.1</span> <span>DSL and ATM</span> </div> </a> <ul id="toc-DSL_and_ATM-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Layer_2" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Layer_2"> <div class="vector-toc-text"> <span class="vector-toc-numb">5.2</span> <span>Layer 2</span> </div> </a> <ul id="toc-Layer_2-sublist" class="vector-toc-list"> </ul> </li> </ul> </li> <li id="toc-Performance_metrics" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Performance_metrics"> <div class="vector-toc-text"> <span class="vector-toc-numb">6</span> <span>Performance metrics</span> </div> </a> <ul id="toc-Performance_metrics-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-PSTN_integration" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#PSTN_integration"> <div class="vector-toc-text"> <span class="vector-toc-numb">7</span> <span>PSTN integration</span> </div> </a> <button aria-controls="toc-PSTN_integration-sublist" class="cdx-button cdx-button--weight-quiet cdx-button--icon-only vector-toc-toggle"> <span class="vector-icon mw-ui-icon-wikimedia-expand"></span> <span>Toggle PSTN integration subsection</span> </button> <ul id="toc-PSTN_integration-sublist" class="vector-toc-list"> <li id="toc-Number_portability" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Number_portability"> <div class="vector-toc-text"> <span class="vector-toc-numb">7.1</span> <span>Number portability</span> </div> </a> <ul id="toc-Number_portability-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Emergency_calls" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Emergency_calls"> <div class="vector-toc-text"> <span class="vector-toc-numb">7.2</span> <span>Emergency calls</span> </div> </a> <ul id="toc-Emergency_calls-sublist" class="vector-toc-list"> </ul> </li> </ul> </li> <li id="toc-Fax_support" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Fax_support"> <div class="vector-toc-text"> <span class="vector-toc-numb">8</span> <span>Fax support</span> </div> </a> <ul id="toc-Fax_support-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Power_requirements" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Power_requirements"> <div class="vector-toc-text"> <span class="vector-toc-numb">9</span> <span>Power requirements</span> </div> </a> <ul id="toc-Power_requirements-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Security" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Security"> <div class="vector-toc-text"> <span class="vector-toc-numb">10</span> <span>Security</span> </div> </a> <ul id="toc-Security-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Caller_ID" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Caller_ID"> <div class="vector-toc-text"> <span class="vector-toc-numb">11</span> <span>Caller ID</span> </div> </a> <ul id="toc-Caller_ID-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Hearing_aid_compatibility" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Hearing_aid_compatibility"> <div class="vector-toc-text"> <span class="vector-toc-numb">12</span> <span>Hearing aid compatibility</span> </div> </a> <ul id="toc-Hearing_aid_compatibility-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Operational_cost" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Operational_cost"> <div class="vector-toc-text"> <span class="vector-toc-numb">13</span> <span>Operational cost</span> </div> </a> <ul id="toc-Operational_cost-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Regulatory_and_legal_issues" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Regulatory_and_legal_issues"> <div class="vector-toc-text"> <span class="vector-toc-numb">14</span> <span>Regulatory and legal issues</span> </div> </a> <button aria-controls="toc-Regulatory_and_legal_issues-sublist" class="cdx-button cdx-button--weight-quiet cdx-button--icon-only vector-toc-toggle"> <span class="vector-icon mw-ui-icon-wikimedia-expand"></span> <span>Toggle Regulatory and legal issues subsection</span> </button> <ul id="toc-Regulatory_and_legal_issues-sublist" class="vector-toc-list"> <li id="toc-Canada" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Canada"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.1</span> <span>Canada</span> </div> </a> <ul id="toc-Canada-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-European_Union" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#European_Union"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.2</span> <span>European Union</span> </div> </a> <ul id="toc-European_Union-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Arab_states_of_the_GCC" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Arab_states_of_the_GCC"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.3</span> <span>Arab states of the GCC</span> </div> </a> <ul id="toc-Arab_states_of_the_GCC-sublist" class="vector-toc-list"> <li id="toc-Oman" class="vector-toc-list-item vector-toc-level-3"> <a class="vector-toc-link" href="#Oman"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.3.1</span> <span>Oman</span> </div> </a> <ul id="toc-Oman-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Saudi_Arabia" class="vector-toc-list-item vector-toc-level-3"> <a class="vector-toc-link" href="#Saudi_Arabia"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.3.2</span> <span>Saudi Arabia</span> </div> </a> <ul id="toc-Saudi_Arabia-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-United_Arab_Emirates" class="vector-toc-list-item vector-toc-level-3"> <a class="vector-toc-link" href="#United_Arab_Emirates"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.3.3</span> <span>United Arab Emirates</span> </div> </a> <ul id="toc-United_Arab_Emirates-sublist" class="vector-toc-list"> </ul> </li> </ul> </li> <li id="toc-India" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#India"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.4</span> <span>India</span> </div> </a> <ul id="toc-India-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-South_Korea" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#South_Korea"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.5</span> <span>South Korea</span> </div> </a> <ul id="toc-South_Korea-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-United_States" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#United_States"> <div class="vector-toc-text"> <span class="vector-toc-numb">14.6</span> <span>United States</span> </div> </a> <ul id="toc-United_States-sublist" class="vector-toc-list"> </ul> </li> </ul> </li> <li id="toc-History" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#History"> <div class="vector-toc-text"> <span class="vector-toc-numb">15</span> <span>History</span> </div> </a> <button aria-controls="toc-History-sublist" class="cdx-button cdx-button--weight-quiet cdx-button--icon-only vector-toc-toggle"> <span class="vector-icon mw-ui-icon-wikimedia-expand"></span> <span>Toggle History subsection</span> </button> <ul id="toc-History-sublist" class="vector-toc-list"> <li id="toc-Milestones" class="vector-toc-list-item vector-toc-level-2"> <a class="vector-toc-link" href="#Milestones"> <div class="vector-toc-text"> <span class="vector-toc-numb">15.1</span> <span>Milestones</span> </div> </a> <ul id="toc-Milestones-sublist" class="vector-toc-list"> </ul> </li> </ul> </li> <li id="toc-See_also" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#See_also"> <div class="vector-toc-text"> <span class="vector-toc-numb">16</span> <span>See also</span> </div> </a> <ul id="toc-See_also-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-Notes" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#Notes"> <div class="vector-toc-text"> <span class="vector-toc-numb">17</span> <span>Notes</span> </div> </a> <ul id="toc-Notes-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-References" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#References"> <div class="vector-toc-text"> <span class="vector-toc-numb">18</span> <span>References</span> </div> </a> <ul id="toc-References-sublist" class="vector-toc-list"> </ul> </li> <li id="toc-External_links" class="vector-toc-list-item vector-toc-level-1"> <a class="vector-toc-link" href="#External_links"> <div class="vector-toc-text"> <span class="vector-toc-numb">19</span> <span>External links</span> </div> </a> <ul id="toc-External_links-sublist" class="vector-toc-list"> </ul> </li> </ul> </div> </div> </nav> </div> </div> <div class="mw-content-container"> <main id="content" class="mw-body"> <header class="mw-body-header vector-page-titlebar"> <nav aria-label="Contents" class="vector-toc-landmark"> <div id="vector-page-titlebar-toc" class="vector-dropdown vector-page-titlebar-toc vector-button-flush-left" > <input type="checkbox" id="vector-page-titlebar-toc-checkbox" role="button" aria-haspopup="true" data-event-name="ui.dropdown-vector-page-titlebar-toc" class="vector-dropdown-checkbox " aria-label="Toggle the table of contents" > <label id="vector-page-titlebar-toc-label" for="vector-page-titlebar-toc-checkbox" class="vector-dropdown-label cdx-button cdx-button--fake-button cdx-button--fake-button--enabled cdx-button--weight-quiet cdx-button--icon-only " aria-hidden="true" ><span class="vector-icon mw-ui-icon-listBullet mw-ui-icon-wikimedia-listBullet"></span> <span class="vector-dropdown-label-text">Toggle the table of contents</span> </label> <div class="vector-dropdown-content"> <div id="vector-page-titlebar-toc-unpinned-container" class="vector-unpinned-container"> </div> </div> </div> </nav> <h1 id="firstHeading" class="firstHeading mw-first-heading"><span class="mw-page-title-main">Voice over IP</span></h1> <div id="p-lang-btn" class="vector-dropdown mw-portlet mw-portlet-lang" > <input type="checkbox" id="p-lang-btn-checkbox" role="button" aria-haspopup="true" data-event-name="ui.dropdown-p-lang-btn" class="vector-dropdown-checkbox mw-interlanguage-selector" aria-label="Go to an article in another language. Available in 73 languages" > <label id="p-lang-btn-label" for="p-lang-btn-checkbox" class="vector-dropdown-label cdx-button cdx-button--fake-button cdx-button--fake-button--enabled cdx-button--weight-quiet cdx-button--action-progressive mw-portlet-lang-heading-73" aria-hidden="true" ><span class="vector-icon mw-ui-icon-language-progressive mw-ui-icon-wikimedia-language-progressive"></span> <span class="vector-dropdown-label-text">73 languages</span> </label> <div class="vector-dropdown-content"> <div class="vector-menu-content"> <ul class="vector-menu-content-list"> <li class="interlanguage-link interwiki-af mw-list-item"><a href="https://af.wikipedia.org/wiki/Stem_oor_IP" title="Stem oor IP – Afrikaans" lang="af" hreflang="af" data-title="Stem oor IP" data-language-autonym="Afrikaans" data-language-local-name="Afrikaans" class="interlanguage-link-target"><span>Afrikaans</span></a></li><li class="interlanguage-link interwiki-ar mw-list-item"><a href="https://ar.wikipedia.org/wiki/%D9%86%D9%82%D9%84_%D8%A7%D9%84%D8%B5%D9%88%D8%AA_%D8%A8%D8%A7%D8%B3%D8%AA%D8%B9%D9%85%D8%A7%D9%84_%D8%A8%D8%B1%D9%88%D8%AA%D9%88%D9%83%D9%88%D9%84_%D8%A7%D9%84%D8%A5%D9%86%D8%AA%D8%B1%D9%86%D8%AA" title="نقل الصوت باستعمال بروتوكول الإنترنت – Arabic" lang="ar" hreflang="ar" data-title="نقل الصوت باستعمال بروتوكول الإنترنت" data-language-autonym="العربية" data-language-local-name="Arabic" class="interlanguage-link-target"><span>العربية</span></a></li><li class="interlanguage-link interwiki-ast mw-list-item"><a href="https://ast.wikipedia.org/wiki/Voz_sobre_protocolu_d%27internet" title="Voz sobre protocolu d'internet – Asturian" lang="ast" hreflang="ast" data-title="Voz sobre protocolu d'internet" data-language-autonym="Asturianu" data-language-local-name="Asturian" class="interlanguage-link-target"><span>Asturianu</span></a></li><li class="interlanguage-link interwiki-az mw-list-item"><a href="https://az.wikipedia.org/wiki/IP-telefoniya" title="IP-telefoniya – Azerbaijani" lang="az" hreflang="az" data-title="IP-telefoniya" data-language-autonym="Azərbaycanca" data-language-local-name="Azerbaijani" class="interlanguage-link-target"><span>Azərbaycanca</span></a></li><li class="interlanguage-link interwiki-bn mw-list-item"><a href="https://bn.wikipedia.org/wiki/%E0%A6%AD%E0%A6%AF%E0%A6%BC%E0%A7%87%E0%A6%B8_%E0%A6%93%E0%A6%AD%E0%A6%BE%E0%A6%B0_%E0%A6%87%E0%A6%A8%E0%A7%8D%E0%A6%9F%E0%A6%BE%E0%A6%B0%E0%A6%A8%E0%A7%87%E0%A6%9F_%E0%A6%AA%E0%A7%8D%E0%A6%B0%E0%A7%8B%E0%A6%9F%E0%A7%8B%E0%A6%95%E0%A6%B2" title="ভয়েস ওভার ইন্টারনেট প্রোটোকল – Bangla" lang="bn" hreflang="bn" data-title="ভয়েস ওভার ইন্টারনেট প্রোটোকল" data-language-autonym="বাংলা" data-language-local-name="Bangla" class="interlanguage-link-target"><span>বাংলা</span></a></li><li class="interlanguage-link interwiki-zh-min-nan mw-list-item"><a href="https://zh-min-nan.wikipedia.org/wiki/VoIP" title="VoIP – Minnan" lang="nan" hreflang="nan" data-title="VoIP" data-language-autonym="閩南語 / Bân-lâm-gú" data-language-local-name="Minnan" class="interlanguage-link-target"><span>閩南語 / Bân-lâm-gú</span></a></li><li class="interlanguage-link interwiki-ba mw-list-item"><a href="https://ba.wikipedia.org/wiki/IP-%D1%82%D0%B5%D0%BB%D0%B5%D1%84%D0%BE%D0%BD%D0%B8%D1%8F" title="IP-телефония – Bashkir" lang="ba" hreflang="ba" data-title="IP-телефония" data-language-autonym="Башҡортса" data-language-local-name="Bashkir" class="interlanguage-link-target"><span>Башҡортса</span></a></li><li class="interlanguage-link interwiki-be mw-list-item"><a href="https://be.wikipedia.org/wiki/VoIP" title="VoIP – Belarusian" lang="be" hreflang="be" data-title="VoIP" data-language-autonym="Беларуская" data-language-local-name="Belarusian" class="interlanguage-link-target"><span>Беларуская</span></a></li><li class="interlanguage-link interwiki-bg mw-list-item"><a href="https://bg.wikipedia.org/wiki/%D0%98%D0%BD%D1%82%D0%B5%D1%80%D0%BD%D0%B5%D1%82_%D1%82%D0%B5%D0%BB%D0%B5%D1%84%D0%BE%D0%BD%D0%B8%D1%8F" title="Интернет телефония – Bulgarian" lang="bg" hreflang="bg" data-title="Интернет телефония" data-language-autonym="Български" data-language-local-name="Bulgarian" class="interlanguage-link-target"><span>Български</span></a></li><li class="interlanguage-link interwiki-bs mw-list-item"><a href="https://bs.wikipedia.org/wiki/Voice_over_Internet_Protocol" title="Voice over Internet Protocol – Bosnian" lang="bs" hreflang="bs" data-title="Voice over Internet Protocol" data-language-autonym="Bosanski" data-language-local-name="Bosnian" class="interlanguage-link-target"><span>Bosanski</span></a></li><li class="interlanguage-link interwiki-ca mw-list-item"><a href="https://ca.wikipedia.org/wiki/Veu_per_IP" title="Veu per IP – Catalan" lang="ca" hreflang="ca" data-title="Veu per IP" data-language-autonym="Català" data-language-local-name="Catalan" class="interlanguage-link-target"><span>Català</span></a></li><li class="interlanguage-link interwiki-cs mw-list-item"><a href="https://cs.wikipedia.org/wiki/Voice_over_Internet_Protocol" title="Voice over Internet Protocol – Czech" lang="cs" hreflang="cs" data-title="Voice over Internet Protocol" data-language-autonym="Čeština" data-language-local-name="Czech" class="interlanguage-link-target"><span>Čeština</span></a></li><li class="interlanguage-link interwiki-da mw-list-item"><a href="https://da.wikipedia.org/wiki/VoIP" title="VoIP – Danish" lang="da" hreflang="da" data-title="VoIP" data-language-autonym="Dansk" data-language-local-name="Danish" class="interlanguage-link-target"><span>Dansk</span></a></li><li class="interlanguage-link interwiki-de mw-list-item"><a href="https://de.wikipedia.org/wiki/IP-Telefonie" title="IP-Telefonie – German" lang="de" hreflang="de" data-title="IP-Telefonie" data-language-autonym="Deutsch" data-language-local-name="German" class="interlanguage-link-target"><span>Deutsch</span></a></li><li class="interlanguage-link interwiki-et mw-list-item"><a href="https://et.wikipedia.org/wiki/Internetitelefon" title="Internetitelefon – Estonian" lang="et" hreflang="et" data-title="Internetitelefon" data-language-autonym="Eesti" data-language-local-name="Estonian" class="interlanguage-link-target"><span>Eesti</span></a></li><li class="interlanguage-link interwiki-el mw-list-item"><a href="https://el.wikipedia.org/wiki/VoIP" title="VoIP – Greek" lang="el" hreflang="el" data-title="VoIP" data-language-autonym="Ελληνικά" data-language-local-name="Greek" class="interlanguage-link-target"><span>Ελληνικά</span></a></li><li class="interlanguage-link interwiki-es mw-list-item"><a href="https://es.wikipedia.org/wiki/Voz_sobre_protocolo_de_internet" title="Voz sobre protocolo de internet – Spanish" lang="es" hreflang="es" data-title="Voz sobre protocolo de internet" data-language-autonym="Español" data-language-local-name="Spanish" class="interlanguage-link-target"><span>Español</span></a></li><li class="interlanguage-link interwiki-eo mw-list-item"><a href="https://eo.wikipedia.org/wiki/Interreta_telefono" title="Interreta telefono – Esperanto" lang="eo" hreflang="eo" data-title="Interreta telefono" data-language-autonym="Esperanto" data-language-local-name="Esperanto" class="interlanguage-link-target"><span>Esperanto</span></a></li><li class="interlanguage-link interwiki-eu mw-list-item"><a href="https://eu.wikipedia.org/wiki/VoIP" title="VoIP – Basque" lang="eu" hreflang="eu" data-title="VoIP" data-language-autonym="Euskara" data-language-local-name="Basque" class="interlanguage-link-target"><span>Euskara</span></a></li><li class="interlanguage-link interwiki-fa mw-list-item"><a href="https://fa.wikipedia.org/wiki/%D8%B5%D8%AF%D8%A7_%D8%B1%D9%88%DB%8C_%D9%BE%D8%B1%D9%88%D8%AA%DA%A9%D9%84_%D8%A7%DB%8C%D9%86%D8%AA%D8%B1%D9%86%D8%AA" title="صدا روی پروتکل اینترنت – Persian" lang="fa" hreflang="fa" data-title="صدا روی پروتکل اینترنت" data-language-autonym="فارسی" data-language-local-name="Persian" class="interlanguage-link-target"><span>فارسی</span></a></li><li class="interlanguage-link interwiki-fr mw-list-item"><a href="https://fr.wikipedia.org/wiki/Voix_sur_IP" title="Voix sur IP – French" lang="fr" hreflang="fr" data-title="Voix sur IP" data-language-autonym="Français" data-language-local-name="French" class="interlanguage-link-target"><span>Français</span></a></li><li class="interlanguage-link interwiki-ga mw-list-item"><a href="https://ga.wikipedia.org/wiki/VoIP" title="VoIP – Irish" lang="ga" hreflang="ga" data-title="VoIP" data-language-autonym="Gaeilge" data-language-local-name="Irish" class="interlanguage-link-target"><span>Gaeilge</span></a></li><li class="interlanguage-link interwiki-gl mw-list-item"><a href="https://gl.wikipedia.org/wiki/VoIP" title="VoIP – Galician" lang="gl" hreflang="gl" data-title="VoIP" data-language-autonym="Galego" data-language-local-name="Galician" class="interlanguage-link-target"><span>Galego</span></a></li><li class="interlanguage-link interwiki-ko mw-list-item"><a href="https://ko.wikipedia.org/wiki/%EC%9D%8C%EC%84%B1_%EC%9D%B8%ED%84%B0%EB%84%B7_%ED%94%84%EB%A1%9C%ED%86%A0%EC%BD%9C" title="음성 인터넷 프로토콜 – Korean" lang="ko" hreflang="ko" data-title="음성 인터넷 프로토콜" data-language-autonym="한국어" data-language-local-name="Korean" class="interlanguage-link-target"><span>한국어</span></a></li><li class="interlanguage-link interwiki-hi mw-list-item"><a href="https://hi.wikipedia.org/wiki/%E0%A4%B5%E0%A5%89%E0%A4%87%E0%A4%B8_%E0%A4%93%E0%A4%B5%E0%A4%B0_%E0%A4%87%E0%A4%A3%E0%A5%8D%E0%A4%9F%E0%A4%B0%E0%A4%A8%E0%A5%87%E0%A4%9F_%E0%A4%AA%E0%A5%8D%E0%A4%B0%E0%A5%8B%E0%A4%9F%E0%A5%8B%E0%A4%95%E0%A5%89%E0%A4%B2" title="वॉइस ओवर इण्टरनेट प्रोटोकॉल – Hindi" lang="hi" hreflang="hi" data-title="वॉइस ओवर इण्टरनेट प्रोटोकॉल" data-language-autonym="हिन्दी" data-language-local-name="Hindi" class="interlanguage-link-target"><span>हिन्दी</span></a></li><li class="interlanguage-link interwiki-hr mw-list-item"><a href="https://hr.wikipedia.org/wiki/VoIP" title="VoIP – Croatian" lang="hr" hreflang="hr" data-title="VoIP" data-language-autonym="Hrvatski" data-language-local-name="Croatian" class="interlanguage-link-target"><span>Hrvatski</span></a></li><li class="interlanguage-link interwiki-io mw-list-item"><a href="https://io.wikipedia.org/wiki/IP-telefonado" title="IP-telefonado – Ido" lang="io" hreflang="io" data-title="IP-telefonado" data-language-autonym="Ido" data-language-local-name="Ido" class="interlanguage-link-target"><span>Ido</span></a></li><li class="interlanguage-link interwiki-id mw-list-item"><a href="https://id.wikipedia.org/wiki/Voice_over_IP" title="Voice over IP – Indonesian" lang="id" hreflang="id" data-title="Voice over IP" data-language-autonym="Bahasa Indonesia" data-language-local-name="Indonesian" class="interlanguage-link-target"><span>Bahasa Indonesia</span></a></li><li class="interlanguage-link interwiki-xh mw-list-item"><a href="https://xh.wikipedia.org/wiki/Ilizwi_nge-IP" title="Ilizwi nge-IP – Xhosa" lang="xh" hreflang="xh" data-title="Ilizwi nge-IP" data-language-autonym="IsiXhosa" data-language-local-name="Xhosa" class="interlanguage-link-target"><span>IsiXhosa</span></a></li><li class="interlanguage-link interwiki-it mw-list-item"><a href="https://it.wikipedia.org/wiki/Voice_over_IP" title="Voice over IP – Italian" lang="it" hreflang="it" data-title="Voice over IP" data-language-autonym="Italiano" data-language-local-name="Italian" class="interlanguage-link-target"><span>Italiano</span></a></li><li class="interlanguage-link interwiki-he mw-list-item"><a href="https://he.wikipedia.org/wiki/VoIP" title="VoIP – Hebrew" lang="he" hreflang="he" data-title="VoIP" data-language-autonym="עברית" data-language-local-name="Hebrew" class="interlanguage-link-target"><span>עברית</span></a></li><li class="interlanguage-link interwiki-kn mw-list-item"><a href="https://kn.wikipedia.org/wiki/%E0%B2%85%E0%B2%82%E0%B2%A4%E0%B2%B0%E0%B2%9C%E0%B2%BE%E0%B2%B2_%E0%B2%86%E0%B2%A7%E0%B2%BE%E0%B2%B0%E0%B2%BF%E0%B2%A4_%E0%B2%95%E0%B2%B0%E0%B3%86_%E0%B2%AA%E0%B3%8D%E0%B2%B0%E0%B3%8B%E0%B2%9F%E0%B3%8B%E0%B2%95%E0%B2%BE%E0%B2%B2%E0%B3%8D%E2%80%8C" title="ಅಂತರಜಾಲ ಆಧಾರಿತ ಕರೆ ಪ್ರೋಟೋಕಾಲ್ – Kannada" lang="kn" hreflang="kn" data-title="ಅಂತರಜಾಲ ಆಧಾರಿತ ಕರೆ ಪ್ರೋಟೋಕಾಲ್" data-language-autonym="ಕನ್ನಡ" data-language-local-name="Kannada" class="interlanguage-link-target"><span>ಕನ್ನಡ</span></a></li><li class="interlanguage-link interwiki-kk mw-list-item"><a href="https://kk.wikipedia.org/wiki/IP-%D1%82%D0%B5%D0%BB%D0%B5%D1%84%D0%BE%D0%BD%D0%B8%D1%8F" title="IP-телефония – Kazakh" lang="kk" hreflang="kk" data-title="IP-телефония" data-language-autonym="Қазақша" data-language-local-name="Kazakh" class="interlanguage-link-target"><span>Қазақша</span></a></li><li class="interlanguage-link interwiki-lo mw-list-item"><a href="https://lo.wikipedia.org/wiki/VoIP" title="VoIP – Lao" lang="lo" hreflang="lo" data-title="VoIP" data-language-autonym="ລາວ" data-language-local-name="Lao" class="interlanguage-link-target"><span>ລາວ</span></a></li><li class="interlanguage-link interwiki-lv mw-list-item"><a href="https://lv.wikipedia.org/wiki/IP_balss_p%C4%81rraide" title="IP balss pārraide – Latvian" lang="lv" hreflang="lv" data-title="IP balss pārraide" data-language-autonym="Latviešu" data-language-local-name="Latvian" class="interlanguage-link-target"><span>Latviešu</span></a></li><li class="interlanguage-link interwiki-lt mw-list-item"><a href="https://lt.wikipedia.org/wiki/IP_telefonija" title="IP telefonija – Lithuanian" lang="lt" hreflang="lt" data-title="IP telefonija" data-language-autonym="Lietuvių" data-language-local-name="Lithuanian" class="interlanguage-link-target"><span>Lietuvių</span></a></li><li class="interlanguage-link interwiki-lmo mw-list-item"><a href="https://lmo.wikipedia.org/wiki/Vus_sura_IP" title="Vus sura IP – Lombard" lang="lmo" hreflang="lmo" data-title="Vus sura IP" data-language-autonym="Lombard" data-language-local-name="Lombard" class="interlanguage-link-target"><span>Lombard</span></a></li><li class="interlanguage-link interwiki-hu mw-list-item"><a href="https://hu.wikipedia.org/wiki/Voice_over_IP" title="Voice over IP – Hungarian" lang="hu" hreflang="hu" data-title="Voice over IP" data-language-autonym="Magyar" data-language-local-name="Hungarian" class="interlanguage-link-target"><span>Magyar</span></a></li><li class="interlanguage-link interwiki-mk mw-list-item"><a href="https://mk.wikipedia.org/wiki/Voice_over_IP" title="Voice over IP – Macedonian" lang="mk" hreflang="mk" data-title="Voice over IP" data-language-autonym="Македонски" data-language-local-name="Macedonian" class="interlanguage-link-target"><span>Македонски</span></a></li><li class="interlanguage-link interwiki-ml mw-list-item"><a href="https://ml.wikipedia.org/wiki/%E0%B4%B5%E0%B5%8B%E0%B4%AF%E0%B4%BF%E0%B4%B8%E0%B5%8D_%E0%B4%93%E0%B4%B5%E0%B5%BC_%E0%B4%87%E0%B4%A8%E0%B5%8D%E0%B4%B1%E0%B5%BC%E0%B4%A8%E0%B5%86%E0%B4%B1%E0%B5%8D%E0%B4%B1%E0%B5%8D_%E0%B4%AA%E0%B5%8D%E0%B4%B0%E0%B5%8B%E0%B4%9F%E0%B5%8D%E0%B4%9F%E0%B5%8B%E0%B4%95%E0%B5%8D%E0%B4%95%E0%B5%8B%E0%B5%BE" title="വോയിസ് ഓവർ ഇന്റർനെറ്റ് പ്രോട്ടോക്കോൾ – Malayalam" lang="ml" hreflang="ml" data-title="വോയിസ് ഓവർ ഇന്റർനെറ്റ് പ്രോട്ടോക്കോൾ" data-language-autonym="മലയാളം" data-language-local-name="Malayalam" class="interlanguage-link-target"><span>മലയാളം</span></a></li><li class="interlanguage-link interwiki-ms mw-list-item"><a href="https://ms.wikipedia.org/wiki/Suara_melalui_Protokol_Internet" title="Suara melalui Protokol Internet – Malay" lang="ms" hreflang="ms" data-title="Suara melalui Protokol Internet" data-language-autonym="Bahasa Melayu" data-language-local-name="Malay" class="interlanguage-link-target"><span>Bahasa Melayu</span></a></li><li class="interlanguage-link interwiki-my mw-list-item"><a href="https://my.wikipedia.org/wiki/%E1%80%97%E1%80%BD%E1%80%AE%E1%80%A1%E1%80%AD%E1%80%AF%E1%80%A1%E1%80%AD%E1%80%AF%E1%80%84%E1%80%BA%E1%80%95%E1%80%AE" title="ဗွီအိုအိုင်ပီ – Burmese" lang="my" hreflang="my" data-title="ဗွီအိုအိုင်ပီ" data-language-autonym="မြန်မာဘာသာ" data-language-local-name="Burmese" class="interlanguage-link-target"><span>မြန်မာဘာသာ</span></a></li><li class="interlanguage-link interwiki-nl mw-list-item"><a href="https://nl.wikipedia.org/wiki/IP-telefonie" title="IP-telefonie – Dutch" lang="nl" hreflang="nl" data-title="IP-telefonie" data-language-autonym="Nederlands" data-language-local-name="Dutch" class="interlanguage-link-target"><span>Nederlands</span></a></li><li class="interlanguage-link interwiki-ja mw-list-item"><a href="https://ja.wikipedia.org/wiki/VoIP" title="VoIP – Japanese" lang="ja" hreflang="ja" data-title="VoIP" data-language-autonym="日本語" data-language-local-name="Japanese" class="interlanguage-link-target"><span>日本語</span></a></li><li class="interlanguage-link interwiki-nn mw-list-item"><a href="https://nn.wikipedia.org/wiki/IP-telefoni" title="IP-telefoni – Norwegian Nynorsk" lang="nn" hreflang="nn" data-title="IP-telefoni" data-language-autonym="Norsk nynorsk" data-language-local-name="Norwegian Nynorsk" class="interlanguage-link-target"><span>Norsk nynorsk</span></a></li><li class="interlanguage-link interwiki-pms mw-list-item"><a href="https://pms.wikipedia.org/wiki/Telefon%C3%ACa_IP" title="Telefonìa IP – Piedmontese" lang="pms" hreflang="pms" data-title="Telefonìa IP" data-language-autonym="Piemontèis" data-language-local-name="Piedmontese" class="interlanguage-link-target"><span>Piemontèis</span></a></li><li class="interlanguage-link interwiki-pl mw-list-item"><a href="https://pl.wikipedia.org/wiki/Voice_over_Internet_Protocol" title="Voice over Internet Protocol – Polish" lang="pl" hreflang="pl" data-title="Voice over Internet Protocol" data-language-autonym="Polski" data-language-local-name="Polish" class="interlanguage-link-target"><span>Polski</span></a></li><li class="interlanguage-link interwiki-pt mw-list-item"><a href="https://pt.wikipedia.org/wiki/Voz_sobre_IP" title="Voz sobre IP – Portuguese" lang="pt" hreflang="pt" data-title="Voz sobre IP" data-language-autonym="Português" data-language-local-name="Portuguese" class="interlanguage-link-target"><span>Português</span></a></li><li class="interlanguage-link interwiki-ro mw-list-item"><a href="https://ro.wikipedia.org/wiki/Voce_peste_IP" title="Voce peste IP – Romanian" lang="ro" hreflang="ro" data-title="Voce peste IP" data-language-autonym="Română" data-language-local-name="Romanian" class="interlanguage-link-target"><span>Română</span></a></li><li class="interlanguage-link interwiki-qu mw-list-item"><a href="https://qu.wikipedia.org/wiki/IP_patapi_kunka" title="IP patapi kunka – Quechua" lang="qu" hreflang="qu" data-title="IP patapi kunka" data-language-autonym="Runa Simi" data-language-local-name="Quechua" class="interlanguage-link-target"><span>Runa Simi</span></a></li><li class="interlanguage-link interwiki-ru mw-list-item"><a href="https://ru.wikipedia.org/wiki/IP-%D1%82%D0%B5%D0%BB%D0%B5%D1%84%D0%BE%D0%BD%D0%B8%D1%8F" title="IP-телефония – Russian" lang="ru" hreflang="ru" data-title="IP-телефония" data-language-autonym="Русский" data-language-local-name="Russian" class="interlanguage-link-target"><span>Русский</span></a></li><li class="interlanguage-link interwiki-sq mw-list-item"><a href="https://sq.wikipedia.org/wiki/Z%C3%ABri_p%C3%ABrmes_IP" title="Zëri përmes IP – Albanian" lang="sq" hreflang="sq" data-title="Zëri përmes IP" data-language-autonym="Shqip" data-language-local-name="Albanian" class="interlanguage-link-target"><span>Shqip</span></a></li><li class="interlanguage-link interwiki-si mw-list-item"><a href="https://si.wikipedia.org/wiki/%E0%B6%85%E0%B6%B1%E0%B7%8A%E0%B6%AD%E0%B6%BB%E0%B7%8A%E0%B6%A2%E0%B7%8F%E0%B6%BD_%E0%B6%B4%E0%B7%8A%E2%80%8D%E0%B6%BB%E0%B7%9C%E0%B6%A7%E0%B7%9D%E0%B6%9A%E0%B7%9D%E0%B6%BD%E0%B6%BA_%E0%B7%84%E0%B6%BB%E0%B7%84%E0%B7%8F_%E0%B6%9A%E0%B6%A7%E0%B7%84%E0%B6%AC" title="අන්තර්ජාල ප්රොටෝකෝලය හරහා කටහඬ – Sinhala" lang="si" hreflang="si" data-title="අන්තර්ජාල ප්රොටෝකෝලය හරහා කටහඬ" data-language-autonym="සිංහල" data-language-local-name="Sinhala" class="interlanguage-link-target"><span>සිංහල</span></a></li><li class="interlanguage-link interwiki-simple mw-list-item"><a href="https://simple.wikipedia.org/wiki/Voice_over_Internet_Protocol" title="Voice over Internet Protocol – Simple English" lang="en-simple" hreflang="en-simple" data-title="Voice over Internet Protocol" data-language-autonym="Simple English" data-language-local-name="Simple English" class="interlanguage-link-target"><span>Simple English</span></a></li><li class="interlanguage-link interwiki-sk mw-list-item"><a href="https://sk.wikipedia.org/wiki/Prenos_hlasu_internetov%C3%BDm_protokolom" title="Prenos hlasu internetovým protokolom – Slovak" lang="sk" hreflang="sk" data-title="Prenos hlasu internetovým protokolom" data-language-autonym="Slovenčina" data-language-local-name="Slovak" class="interlanguage-link-target"><span>Slovenčina</span></a></li><li class="interlanguage-link interwiki-sl mw-list-item"><a href="https://sl.wikipedia.org/wiki/VoIP_telefonija" title="VoIP telefonija – Slovenian" lang="sl" hreflang="sl" data-title="VoIP telefonija" data-language-autonym="Slovenščina" data-language-local-name="Slovenian" class="interlanguage-link-target"><span>Slovenščina</span></a></li><li class="interlanguage-link interwiki-so mw-list-item"><a href="https://so.wikipedia.org/wiki/VOIP" title="VOIP – Somali" lang="so" hreflang="so" data-title="VOIP" data-language-autonym="Soomaaliga" data-language-local-name="Somali" class="interlanguage-link-target"><span>Soomaaliga</span></a></li><li class="interlanguage-link interwiki-sr mw-list-item"><a href="https://sr.wikipedia.org/wiki/%D0%98%D0%BD%D1%82%D0%B5%D1%80%D0%BD%D0%B5%D1%82_%D1%82%D0%B5%D0%BB%D0%B5%D1%84%D0%BE%D0%BD%D0%B8%D1%98%D0%B0" title="Интернет телефонија – Serbian" lang="sr" hreflang="sr" data-title="Интернет телефонија" data-language-autonym="Српски / srpski" data-language-local-name="Serbian" class="interlanguage-link-target"><span>Српски / srpski</span></a></li><li class="interlanguage-link interwiki-su mw-list-item"><a href="https://su.wikipedia.org/wiki/Voice_over_Internet_Protocol" title="Voice over Internet Protocol – Sundanese" lang="su" hreflang="su" data-title="Voice over Internet Protocol" data-language-autonym="Sunda" data-language-local-name="Sundanese" class="interlanguage-link-target"><span>Sunda</span></a></li><li class="interlanguage-link interwiki-fi mw-list-item"><a href="https://fi.wikipedia.org/wiki/VoIP" title="VoIP – Finnish" lang="fi" hreflang="fi" data-title="VoIP" data-language-autonym="Suomi" data-language-local-name="Finnish" class="interlanguage-link-target"><span>Suomi</span></a></li><li class="interlanguage-link interwiki-sv mw-list-item"><a href="https://sv.wikipedia.org/wiki/IP-telefoni" title="IP-telefoni – Swedish" lang="sv" hreflang="sv" data-title="IP-telefoni" data-language-autonym="Svenska" data-language-local-name="Swedish" class="interlanguage-link-target"><span>Svenska</span></a></li><li class="interlanguage-link interwiki-ta mw-list-item"><a href="https://ta.wikipedia.org/wiki/%E0%AE%87%E0%AE%A3%E0%AF%88%E0%AE%AF%E0%AE%B5%E0%AE%B4%E0%AE%BF_%E0%AE%92%E0%AE%B2%E0%AE%BF_%E0%AE%AA%E0%AE%B0%E0%AE%BF%E0%AE%AE%E0%AE%BE%E0%AE%B1%E0%AF%8D%E0%AE%B1%E0%AE%AE%E0%AF%8D" title="இணையவழி ஒலி பரிமாற்றம் – Tamil" lang="ta" hreflang="ta" data-title="இணையவழி ஒலி பரிமாற்றம்" data-language-autonym="தமிழ்" data-language-local-name="Tamil" class="interlanguage-link-target"><span>தமிழ்</span></a></li><li class="interlanguage-link interwiki-th mw-list-item"><a href="https://th.wikipedia.org/wiki/%E0%B8%A7%E0%B8%AD%E0%B8%A2%E0%B8%8B%E0%B9%8C%E0%B9%82%E0%B8%AD%E0%B9%80%E0%B8%A7%E0%B8%AD%E0%B8%A3%E0%B9%8C%E0%B9%84%E0%B8%AD%E0%B8%9E%E0%B8%B5" title="วอยซ์โอเวอร์ไอพี – Thai" lang="th" hreflang="th" data-title="วอยซ์โอเวอร์ไอพี" data-language-autonym="ไทย" data-language-local-name="Thai" class="interlanguage-link-target"><span>ไทย</span></a></li><li class="interlanguage-link interwiki-tg mw-list-item"><a href="https://tg.wikipedia.org/wiki/IP-%D1%82%D0%B5%D0%BB%D0%B5%D1%84%D0%BE%D0%BD%D0%B8%D1%8F" title="IP-телефония – Tajik" lang="tg" hreflang="tg" data-title="IP-телефония" data-language-autonym="Тоҷикӣ" data-language-local-name="Tajik" class="interlanguage-link-target"><span>Тоҷикӣ</span></a></li><li class="interlanguage-link interwiki-tr mw-list-item"><a href="https://tr.wikipedia.org/wiki/VoIP" title="VoIP – Turkish" lang="tr" hreflang="tr" data-title="VoIP" data-language-autonym="Türkçe" data-language-local-name="Turkish" class="interlanguage-link-target"><span>Türkçe</span></a></li><li class="interlanguage-link interwiki-uk mw-list-item"><a href="https://uk.wikipedia.org/wiki/VoIP" title="VoIP – Ukrainian" lang="uk" hreflang="uk" data-title="VoIP" data-language-autonym="Українська" data-language-local-name="Ukrainian" class="interlanguage-link-target"><span>Українська</span></a></li><li class="interlanguage-link interwiki-ur mw-list-item"><a href="https://ur.wikipedia.org/wiki/%D8%A2%D9%88%D8%A7%D8%B2_%D8%A8%D8%AF%D9%88%D8%B4_%D8%A2%D8%A6%DB%8C_%D9%BE%DB%8C" title="آواز بدوش آئی پی – Urdu" lang="ur" hreflang="ur" data-title="آواز بدوش آئی پی" data-language-autonym="اردو" data-language-local-name="Urdu" class="interlanguage-link-target"><span>اردو</span></a></li><li class="interlanguage-link interwiki-vec mw-list-item"><a href="https://vec.wikipedia.org/wiki/Te%C5%82efono_VoIP" title="Tełefono VoIP – Venetian" lang="vec" hreflang="vec" data-title="Tełefono VoIP" data-language-autonym="Vèneto" data-language-local-name="Venetian" class="interlanguage-link-target"><span>Vèneto</span></a></li><li class="interlanguage-link interwiki-vi mw-list-item"><a href="https://vi.wikipedia.org/wiki/VoIP" title="VoIP – Vietnamese" lang="vi" hreflang="vi" data-title="VoIP" data-language-autonym="Tiếng Việt" data-language-local-name="Vietnamese" class="interlanguage-link-target"><span>Tiếng Việt</span></a></li><li class="interlanguage-link interwiki-wa mw-list-item"><a href="https://wa.wikipedia.org/wiki/Telefoneye_pal_Daegntoele" title="Telefoneye pal Daegntoele – Walloon" lang="wa" hreflang="wa" data-title="Telefoneye pal Daegntoele" data-language-autonym="Walon" data-language-local-name="Walloon" class="interlanguage-link-target"><span>Walon</span></a></li><li class="interlanguage-link interwiki-wuu mw-list-item"><a href="https://wuu.wikipedia.org/wiki/VoIP" title="VoIP – Wu" lang="wuu" hreflang="wuu" data-title="VoIP" data-language-autonym="吴语" data-language-local-name="Wu" class="interlanguage-link-target"><span>吴语</span></a></li><li class="interlanguage-link interwiki-yi mw-list-item"><a href="https://yi.wikipedia.org/wiki/VoIP" title="VoIP – Yiddish" lang="yi" hreflang="yi" data-title="VoIP" data-language-autonym="ייִדיש" data-language-local-name="Yiddish" class="interlanguage-link-target"><span>ייִדיש</span></a></li><li class="interlanguage-link interwiki-zh mw-list-item"><a href="https://zh.wikipedia.org/wiki/VoIP" title="VoIP – Chinese" lang="zh" hreflang="zh" data-title="VoIP" data-language-autonym="中文" data-language-local-name="Chinese" class="interlanguage-link-target"><span>中文</span></a></li> </ul> <div class="after-portlet after-portlet-lang"><span class="wb-langlinks-edit wb-langlinks-link"><a href="https://www.wikidata.org/wiki/Special:EntityPage/Q81945#sitelinks-wikipedia" title="Edit interlanguage links" class="wbc-editpage">Edit links</a></span></div> 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<div class="vector-body-before-content"> <div class="mw-indicators"> </div> <div id="siteSub" class="noprint">From Wikipedia, the free encyclopedia</div> </div> <div id="contentSub"><div id="mw-content-subtitle"></div></div> <div id="mw-content-text" class="mw-body-content"><div class="mw-content-ltr mw-parser-output" lang="en" dir="ltr"><div class="shortdescription nomobile noexcerpt noprint searchaux" style="display:none">Methods of delivering voice communications and multimedia over IP networks</div> <p class="mw-empty-elt"> </p><p><b>Voice over Internet Protocol</b> (<b>VoIP</b>),<sup id="cite_ref-2" class="reference"><a href="#cite_note-2"><span class="cite-bracket">[</span>a<span class="cite-bracket">]</span></a></sup> also known as <b>IP telephony</b>, refers to a set of technologies used for voice communication sessions over <a href="/wiki/Internet_Protocol" title="Internet Protocol">Internet Protocol</a> (IP) networks, such as the <a href="/wiki/Internet" title="Internet">Internet</a>.<sup id="cite_ref-Arora_2023_t525_3-0" class="reference"><a href="#cite_note-Arora_2023_t525-3"><span class="cite-bracket">[</span>2<span class="cite-bracket">]</span></a></sup> VoIP enables <a href="/wiki/Voice_call" class="mw-redirect" title="Voice call">voice calls</a> to be transmitted as data packets, facilitating various methods of <a href="/wiki/Voice_communication" class="mw-redirect" title="Voice communication">voice communication</a>, including traditional applications like Skype, Microsoft Teams, Google Voice, and <a href="/wiki/VoIP_phone" title="VoIP phone">VoIP phones</a>. Regular telephones can also be used for VoIP by connecting them to the Internet via <a href="/wiki/Analog_telephone_adapter" title="Analog telephone adapter">analog telephone adapters</a> (ATAs), which convert traditional telephone signals into digital data packets that can be transmitted over IP networks. </p><p>The broader terms <b>Internet telephony</b>, <b>broadband telephony</b>, and <b>broadband phone service</b> specifically refer to the delivery of voice and other communication services, such as <a href="/wiki/Fax" title="Fax">fax</a>, <a href="/wiki/SMS" title="SMS">SMS</a>, and <a href="/wiki/Voice_messaging" class="mw-redirect" title="Voice messaging">voice messaging</a>, over the Internet, in contrast to the traditional <a href="/wiki/Public_switched_telephone_network" title="Public switched telephone network">public switched telephone network</a> (PSTN), commonly known as <a href="/wiki/Plain_old_telephone_service" title="Plain old telephone service">plain old telephone service</a> (POTS). </p><p>VoIP technology has evolved to integrate with <a href="/wiki/Mobile_telephony" title="Mobile telephony">mobile telephony</a>, including Voice over LTE (<a href="/wiki/VoLTE" class="mw-redirect" title="VoLTE">VoLTE</a>) and Voice over 5G (<a href="/wiki/Vo5G" class="mw-redirect" title="Vo5G">Vo5G</a>), enabling seamless voice communication over <a href="/wiki/Cellular_network" title="Cellular network">mobile data networks</a>. These advancements have extended VoIP's role beyond its traditional use in internet-based applications. It has become a key component of modern mobile infrastructure, as <a href="/wiki/4G" title="4G">4G</a> and <a href="/wiki/5G" title="5G">5G</a> networks rely entirely on this technology for voice transmission. </p> <meta property="mw:PageProp/toc" /> <div class="mw-heading mw-heading2"><h2 id="Overview">Overview</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=1" title="Edit section: Overview"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>The steps and principles involved in originating VoIP telephone calls are similar to traditional digital <a href="/wiki/Telephony" title="Telephony">telephony</a> and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a <a href="/wiki/Circuit-switched_network" class="mw-redirect" title="Circuit-switched network">circuit-switched network</a>, the digital information is packetized and transmission occurs as IP packets over a <a href="/wiki/Packet-switched_network" class="mw-redirect" title="Packet-switched network">packet-switched network</a>. They transport media streams using special media delivery protocols that encode audio and video with <a href="/wiki/Audio_codec" title="Audio codec">audio codecs</a> and <a href="/wiki/Video_codec" title="Video codec">video codecs</a>. Various codecs exist that optimize the media stream based on application requirements and network bandwidth; some implementations rely on <a href="/wiki/Narrowband" title="Narrowband">narrowband</a> and <a href="/wiki/Speech_coding" title="Speech coding">compressed speech</a>, while others support <a href="/wiki/High-fidelity" class="mw-redirect" title="High-fidelity">high-fidelity</a> stereo codecs. </p><p>The most widely used <a href="/wiki/Speech_coding" title="Speech coding">speech coding</a> standards in VoIP are based on the <a href="/wiki/Linear_predictive_coding" title="Linear predictive coding">linear predictive coding</a> (LPC) and <a href="/wiki/Modified_discrete_cosine_transform" title="Modified discrete cosine transform">modified discrete cosine transform</a> (MDCT) compression methods. Popular codecs include the MDCT-based <a href="/wiki/AAC-LD" title="AAC-LD">AAC-LD</a> (used in <a href="/wiki/FaceTime" title="FaceTime">FaceTime</a>), the LPC/MDCT-based <a href="/wiki/Opus_(audio_format)" title="Opus (audio format)">Opus</a> (used in <a href="/wiki/WhatsApp" title="WhatsApp">WhatsApp</a>), the LPC-based <a href="/wiki/SILK" title="SILK">SILK</a> (used in <a href="/wiki/Skype" title="Skype">Skype</a>), <a href="/wiki/%CE%9C-law" class="mw-redirect" title="Μ-law">μ-law</a> and <a href="/wiki/A-law" class="mw-redirect" title="A-law">A-law</a> versions of <a href="/wiki/G.711" title="G.711">G.711</a>, <a href="/wiki/G.722" title="G.722">G.722</a>, and an <a href="/wiki/Open-source_license" title="Open-source license">open source</a> voice codec known as <a href="/wiki/ILBC" class="mw-redirect" title="ILBC">iLBC</a>, a codec that uses only 8 kbit/s each way called <a href="/wiki/G.729" title="G.729">G.729</a>. </p><p>Early providers of voice-over-IP services used business models and offered technical solutions that mirrored the architecture of the legacy telephone network. Second-generation providers, such as <a href="/wiki/Skype" title="Skype">Skype</a>, 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 limited the freedom of users to mix-and-match third-party hardware and software. Third-generation providers, such as <a href="/wiki/Google_Talk" title="Google Talk">Google Talk</a>, adopted the concept of <a href="/wiki/Federated_VoIP" title="Federated VoIP">federated VoIP</a>.<sup id="cite_ref-4" class="reference"><a href="#cite_note-4"><span class="cite-bracket">[</span>3<span class="cite-bracket">]</span></a></sup> These solutions typically allow dynamic interconnection between users in any two domains of the Internet, when a user wishes to place a call. </p><p>In addition to <a href="/wiki/VoIP_phone" title="VoIP phone">VoIP phones</a>, VoIP is also available on many personal computers and other Internet access devices. Calls and SMS text messages may be sent via <a href="/wiki/Wi-Fi" title="Wi-Fi">Wi-Fi</a> or the carrier's <a href="/wiki/Mobile_data" class="mw-redirect" title="Mobile data">mobile data</a> network.<sup id="cite_ref-EBSCOhost_5-0" class="reference"><a href="#cite_note-EBSCOhost-5"><span class="cite-bracket">[</span>4<span class="cite-bracket">]</span></a></sup> VoIP provides a framework for consolidation of all modern communications technologies using a single <a href="/wiki/Unified_communications" title="Unified communications">unified communications</a> system. </p> <div class="mw-heading mw-heading3"><h3 id="Integration_of_VoIP_in_Mobile_Networks">Integration of VoIP in Mobile Networks</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=2" title="Edit section: Integration of VoIP in Mobile Networks"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>VoIP technology has been adapted for use in <a href="/wiki/Cellular_network" title="Cellular network">mobile networks</a>, leading to the development of advanced systems designed to support voice communication over modern data infrastructures. Among these are Voice over LTE (<a href="/wiki/VoLTE" class="mw-redirect" title="VoLTE">VoLTE</a>) and Voice over 5G (<a href="/wiki/Vo5G" class="mw-redirect" title="Vo5G">Vo5G</a>), which enable voice communication over IP-based mobile infrastructures. In contrast to traditional VoIP services, which often function independently of global telephone numbering systems, VoLTE and Vo5G are directly connected to <a href="/wiki/Mobile_operator" class="mw-redirect" title="Mobile operator">mobile operators</a>' infrastructures, providing seamless connectivity to the international telephone network.<sup id="cite_ref-6" class="reference"><a href="#cite_note-6"><span class="cite-bracket">[</span>5<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-7" class="reference"><a href="#cite_note-7"><span class="cite-bracket">[</span>6<span class="cite-bracket">]</span></a></sup> </p><p>VoLTE, introduced as part of <a href="/wiki/4G" title="4G">4G</a> <a href="/wiki/LTE_(telecommunication)" title="LTE (telecommunication)">LTE</a> networks, enables voice communication over an IP-based infrastructure initially developed for data transmission. It offers features such as high-definition voice (<a href="/wiki/HD_Voice" class="mw-redirect" title="HD Voice">HD Voice</a>) and faster call setup times compared to <a href="/wiki/Circuit-switched_network" class="mw-redirect" title="Circuit-switched network">circuit-switched networks</a>.<sup id="cite_ref-8" class="reference"><a href="#cite_note-8"><span class="cite-bracket">[</span>7<span class="cite-bracket">]</span></a></sup> </p><p>Vo5G, the <a href="/wiki/5G" title="5G">5G</a> equivalent of VoLTE, utilizes the increased speed, reduced latency, and greater capacity of 5G networks to further enhance these capabilities.<sup id="cite_ref-9" class="reference"><a href="#cite_note-9"><span class="cite-bracket">[</span>8<span class="cite-bracket">]</span></a></sup> Both VoLTE and Vo5G maintain compatibility with traditional <a href="/wiki/Public_switched_telephone_network" title="Public switched telephone network">public switched telephone networks</a> (PSTNs), allowing users to make and receive calls to and from any <a href="/wiki/Telephone_number" title="Telephone number">telephone number</a> worldwide. </p><p>These technologies differ from standalone VoIP services by being fully integrated with mobile network operators. This integration ensures additional features such as <a href="/wiki/Emergency_call" class="mw-redirect" title="Emergency call">emergency call</a> support and <a href="/wiki/Quality-of-service" class="mw-redirect" title="Quality-of-service">quality-of-service</a> guarantees, making them a central part of modern mobile telecommunication systems. </p> <div class="mw-heading mw-heading2"><h2 id="Protocols">Protocols</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=3" title="Edit section: Protocols"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Voice over IP has been implemented with <a href="/wiki/Proprietary_protocol" title="Proprietary protocol">proprietary protocols</a> and protocols based on <a href="/wiki/Open_standard#Protocols" title="Open standard">open standards</a> in applications such as VoIP phones, mobile applications, and <a href="/wiki/Web-based_VoIP" title="Web-based VoIP">web-based communications</a>. </p><p>A variety of functions are needed to implement VoIP communication. Some protocols perform multiple functions, while others perform only a few and must be used in concert. These functions include: </p> <ul><li><i>Network</i> and <i>transport</i> – Creating reliable transmission over unreliable protocols, which may involve acknowledging receipt of data and retransmitting data that wasn't received.</li> <li><i>Session management</i> – Creating and managing a <a href="/wiki/Session_(computer_science)" title="Session (computer science)">session</a> (sometimes glossed as simply a "call"), which is a connection between two or more peers that provides a context for further communication.</li> <li><i><a href="/wiki/Signaling_(telecommunications)" title="Signaling (telecommunications)">Signaling</a></i> – Performing registration (advertising one's presence and contact information) and discovery (locating someone and obtaining their contact information), dialing (including reporting <a href="/wiki/Call-progress_tone" title="Call-progress tone">call progress</a>), negotiating capabilities, and call control (such as hold, mute, transfer/forwarding, dialing DTMF keys during a call [e.g. to interact with an <a href="/wiki/Automated_attendant" title="Automated attendant">automated attendant</a> or <a href="/wiki/IVR" class="mw-redirect" title="IVR">IVR</a>], etc.).</li> <li><i>Media description</i> – Determining what type of media to send (audio, video, etc.), how to encode/decode it, and how to send/receive it (IP addresses, ports, etc.).</li> <li><i>Media</i> – Transferring the actual media in the call, such as audio, video, text messages, files, etc.</li> <li><i>Quality of service</i> – Providing out-of-band content or feedback about the media such as <a href="/wiki/Lip_sync" title="Lip sync">synchronization</a>, statistics, etc.</li> <li><i>Security</i> – Implementing access control, verifying the identity of other participants (computers or people), and encrypting data to protect the privacy and integrity of the media contents and/or the control messages.</li></ul> <p>VoIP protocols include: </p> <ul><li><a href="/wiki/Matrix_(protocol)" title="Matrix (protocol)">Matrix</a>, open standard for <a href="/wiki/Online_chat" title="Online chat">online chat</a>, voice over IP, and <a href="/wiki/Videotelephony" title="Videotelephony">videotelephony</a></li> <li><a href="/wiki/Session_Initiation_Protocol" title="Session Initiation Protocol">Session Initiation Protocol</a> (SIP),<sup id="cite_ref-10" class="reference"><a href="#cite_note-10"><span class="cite-bracket">[</span>9<span class="cite-bracket">]</span></a></sup> connection management protocol developed by the IETF</li> <li><a href="/wiki/H.323" title="H.323">H.323</a>, one of the first VoIP call signaling and control protocols that found widespread implementation.<sup id="cite_ref-11" class="reference"><a href="#cite_note-11"><span class="cite-bracket">[</span>10<span class="cite-bracket">]</span></a></sup> 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.<sup id="cite_ref-12" class="reference"><a href="#cite_note-12"><span class="cite-bracket">[</span>11<span class="cite-bracket">]</span></a></sup></li> <li><a href="/wiki/Media_Gateway_Control_Protocol" title="Media Gateway Control Protocol">Media Gateway Control Protocol</a> (MGCP), connection management for media gateways</li> <li><a href="/wiki/H.248" title="H.248">H.248</a>, control protocol for media gateways across a converged internetwork consisting of the traditional PSTN and modern packet networks</li> <li><a href="/wiki/Real-time_Transport_Protocol" title="Real-time Transport Protocol">Real-time Transport Protocol</a> (RTP), transport protocol for real-time audio and video data</li> <li><a href="/wiki/Real-time_Transport_Control_Protocol" class="mw-redirect" title="Real-time Transport Control Protocol">Real-time Transport Control Protocol</a> (RTCP), sister protocol for RTP providing stream statistics and status information</li> <li><a href="/wiki/Secure_Real-time_Transport_Protocol" title="Secure Real-time Transport Protocol">Secure Real-time Transport Protocol</a> (SRTP), encrypted version of RTP</li> <li><a href="/wiki/Session_Description_Protocol" title="Session Description Protocol">Session Description Protocol</a> (SDP), a syntax for session initiation and announcement for multi-media communications and <a href="/wiki/WebSocket" title="WebSocket">WebSocket</a> transports.</li> <li><a href="/wiki/Inter-Asterisk_eXchange" title="Inter-Asterisk eXchange">Inter-Asterisk eXchange</a> (IAX), protocol used between <a href="/wiki/Asterisk_PBX" class="mw-redirect" title="Asterisk PBX">Asterisk PBX</a> instances</li> <li><a href="/wiki/Extensible_Messaging_and_Presence_Protocol" class="mw-redirect" title="Extensible Messaging and Presence Protocol">Extensible Messaging and Presence Protocol</a> (XMPP), instant messaging, presence information, and contact list maintenance</li> <li><a href="/wiki/Jingle_(protocol)" title="Jingle (protocol)">Jingle</a>, for peer-to-peer session control in XMPP</li> <li><a href="/wiki/Skype_protocol" title="Skype protocol">Skype protocol</a>, proprietary Internet telephony protocol suite based on peer-to-peer architecture</li></ul> <div class="mw-heading mw-heading2"><h2 id="Adoption">Adoption</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=4" title="Edit section: Adoption"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <div class="mw-heading mw-heading3"><h3 id="Consumer_market">Consumer market</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=5" title="Edit section: Consumer market"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <figure class="mw-default-size" typeof="mw:File/Thumb"><a href="/wiki/File:Voip-typical.gif" class="mw-file-description"><img src="//upload.wikimedia.org/wikipedia/commons/thumb/4/44/Voip-typical.gif/220px-Voip-typical.gif" decoding="async" width="220" height="221" class="mw-file-element" srcset="//upload.wikimedia.org/wikipedia/commons/4/44/Voip-typical.gif 1.5x" data-file-width="300" data-file-height="302" /></a><figcaption>Example of residential network including VoIP</figcaption></figure> <p>Mass-market VoIP services use existing <a href="/wiki/Broadband_Internet_access" class="mw-redirect" title="Broadband Internet access">broadband Internet access</a>, by which subscribers place and receive telephone calls in much the same manner as they would via the PSTN. Full-service VoIP phone companies provide inbound and outbound service with <a href="/wiki/Direct_inbound_dialing" class="mw-redirect" title="Direct inbound dialing">direct inbound dialing</a>. Many offer unlimited domestic calling and sometimes international calls for a flat monthly subscription fee. Phone calls between subscribers of the same provider are usually free when flat-fee service is not available.<sup id="cite_ref-13" class="reference"><a href="#cite_note-13"><span class="cite-bracket">[</span>12<span class="cite-bracket">]</span></a></sup> </p><p>A VoIP phone is necessary to connect to a VoIP service provider. This can be implemented in several ways: </p> <ul><li>Dedicated VoIP phones connect directly to the IP network using technologies such as wired <a href="/wiki/Ethernet" title="Ethernet">Ethernet</a> or <a href="/wiki/Wi-Fi" title="Wi-Fi">Wi-Fi</a>. These are typically designed in the style of traditional digital business telephones.</li> <li>An <a href="/wiki/Analog_telephone_adapter" title="Analog telephone adapter">analog telephone adapter</a> 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 <a href="/wiki/Cable_modem" title="Cable modem">cable modems</a> have this function built in.</li> <li><a href="/wiki/Softphone" title="Softphone">Softphone</a> 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.<sup id="cite_ref-14" class="reference"><a href="#cite_note-14"><span class="cite-bracket">[</span>13<span class="cite-bracket">]</span></a></sup></li></ul> <div class="mw-heading mw-heading3"><h3 id="PSTN_and_mobile_network_providers">PSTN and mobile network providers</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=6" title="Edit section: PSTN and mobile network providers"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>It is increasingly common for telecommunications providers to use VoIP telephony over dedicated and public IP networks as a <a href="/wiki/Backhaul_(telecommunications)" title="Backhaul (telecommunications)">backhaul</a> to connect switching centers and to interconnect with other telephony network providers; this is often referred to as <i>IP backhaul</i>.<sup id="cite_ref-15" class="reference"><a href="#cite_note-15"><span class="cite-bracket">[</span>14<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-16" class="reference"><a href="#cite_note-16"><span class="cite-bracket">[</span>15<span class="cite-bracket">]</span></a></sup> </p><p><a href="/wiki/Smartphones" class="mw-redirect" title="Smartphones">Smartphones</a> may have SIP clients built into the firmware or available as an application download.<sup id="cite_ref-17" class="reference"><a href="#cite_note-17"><span class="cite-bracket">[</span>16<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-18" class="reference"><a href="#cite_note-18"><span class="cite-bracket">[</span>17<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading3"><h3 id="Corporate_use">Corporate use</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=7" title="Edit section: Corporate use"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new <a href="/wiki/Private_branch_exchange" class="mw-redirect" title="Private branch exchange">Private branch exchange</a> (PBX) lines installed internationally were VoIP.<sup id="cite_ref-19" class="reference"><a href="#cite_note-19"><span class="cite-bracket">[</span>18<span class="cite-bracket">]</span></a></sup> For example, in the United States, the <a href="/wiki/Social_Security_Administration" title="Social Security Administration">Social Security Administration</a> is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.<sup id="cite_ref-20" class="reference"><a href="#cite_note-20"><span class="cite-bracket">[</span>19<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-21" class="reference"><a href="#cite_note-21"><span class="cite-bracket">[</span>20<span class="cite-bracket">]</span></a></sup> </p><p>VoIP allows both voice and data communications to be run over a single network, which can significantly reduce infrastructure costs. The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may run on commodity hardware, such as <a href="/wiki/Personal_computer" title="Personal computer">personal computers</a>. Rather than closed architectures, these devices rely on standard interfaces.<sup id="cite_ref-Korzeniowski_22-0" class="reference"><a href="#cite_note-Korzeniowski-22"><span class="cite-bracket">[</span>21<span class="cite-bracket">]</span></a></sup> VoIP devices have simple, intuitive user interfaces, so users can often make simple system configuration changes. Dual-mode phones enable users to continue their conversations as they move between an outside cellular service and an internal <a href="/wiki/Wi-Fi" title="Wi-Fi">Wi-Fi</a> network, so that it is no longer necessary to carry both a desktop phone and a cell phone. Maintenance becomes simpler as there are fewer devices to oversee.<sup id="cite_ref-Korzeniowski_22-1" class="reference"><a href="#cite_note-Korzeniowski-22"><span class="cite-bracket">[</span>21<span class="cite-bracket">]</span></a></sup> </p><p>VoIP solutions aimed at businesses have evolved into <a href="/wiki/Unified_communications" title="Unified communications">unified communications</a> services that treat all communications—phone calls, faxes, voice mail, e-mail, web conferences, and more—as discrete units that can all be delivered via any means and to any handset, including cellphones. Two kinds of service providers are operating in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.<sup id="cite_ref-23" class="reference"><a href="#cite_note-23"><span class="cite-bracket">[</span>22<span class="cite-bracket">]</span></a></sup> </p><p><a href="/wiki/Skype" title="Skype">Skype</a>, which originally marketed itself as a service among friends, has begun to cater to businesses, providing free-of-charge connections between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge.<sup id="cite_ref-24" class="reference"><a href="#cite_note-24"><span class="cite-bracket">[</span>23<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading2"><h2 id="Delivery_mechanisms">Delivery mechanisms</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=8" title="Edit section: Delivery mechanisms"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In general, the provision of VoIP telephony systems to organizational or individual users can be divided into two primary delivery methods: private or on-premises solutions, or externally hosted solutions delivered by third-party providers. On-premises delivery methods are more akin to the classic PBX deployment model for connecting an office to local PSTN networks. </p><p>While many use cases still remain for private or on-premises VoIP systems, the wider market has been gradually shifting toward <i>Cloud</i> or <i>Hosted</i> VoIP solutions. Hosted systems are also generally better suited to smaller or personal use VoIP deployments, where a private system may not be viable for these scenarios. </p> <div class="mw-heading mw-heading3"><h3 id="Hosted_VoIP_systems">Hosted VoIP systems</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=9" title="Edit section: Hosted VoIP systems"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p><i>Hosted</i> or <i>Cloud</i> VoIP solutions involve a service provider or telecommunications carrier hosting the telephone system as a software solution within their own infrastructure. </p><p>Typically this will be one or more <a href="/wiki/Data_center" title="Data center">data centers</a> with geographic relevance to the end-user(s) of the system. This infrastructure is external to the user of the system and is deployed and maintained by the service provider. </p><p>Endpoints, such as VoIP telephones or softphone applications (apps running on a computer or mobile device), will connect to the VoIP service remotely. These connections typically take place over public internet links, such as local fixed WAN breakout or mobile carrier service. </p> <div class="mw-heading mw-heading3"><h3 id="Private_VoIP_systems">Private VoIP systems</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=10" title="Edit section: Private VoIP systems"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <figure class="mw-default-size" typeof="mw:File/Thumb"><a href="/wiki/File:Gigabyte_BRIX_as_Asterisk-based_PBX.jpg" class="mw-file-description"><img src="//upload.wikimedia.org/wikipedia/commons/thumb/5/52/Gigabyte_BRIX_as_Asterisk-based_PBX.jpg/220px-Gigabyte_BRIX_as_Asterisk-based_PBX.jpg" decoding="async" width="220" height="165" class="mw-file-element" srcset="//upload.wikimedia.org/wikipedia/commons/thumb/5/52/Gigabyte_BRIX_as_Asterisk-based_PBX.jpg/330px-Gigabyte_BRIX_as_Asterisk-based_PBX.jpg 1.5x, //upload.wikimedia.org/wikipedia/commons/thumb/5/52/Gigabyte_BRIX_as_Asterisk-based_PBX.jpg/440px-Gigabyte_BRIX_as_Asterisk-based_PBX.jpg 2x" data-file-width="4128" data-file-height="3096" /></a><figcaption><a href="/wiki/Asterisk_(PBX)" title="Asterisk (PBX)">Asterisk</a>-based PBX for small business</figcaption></figure> <p>In the case of a private VoIP system, the primary telephony system itself is located within the private infrastructure of the end-user organization. Usually, the system will be deployed on-premises at a site within the direct control of the organization. This can provide numerous benefits in terms of QoS control (see <a class="mw-selflink-fragment" href="#Quality_of_service">below</a>), cost scalability, and ensuring privacy and security of communications traffic. However, the responsibility for ensuring that the VoIP system remains performant and resilient is predominantly vested in the end-user organization. This is not the case with a Hosted VoIP solution. </p><p>Private VoIP systems can be physical hardware PBX appliances, converged with other infrastructure, or they can be deployed as software applications. Generally, the latter two options will be in the form of a separate virtualized appliance. However, in some scenarios, these systems are deployed on bare metal infrastructure or IoT devices. With some solutions, such as 3CX, companies can attempt to blend the benefits of hosted and private on-premises systems by implementing their own private solution but within an external environment. Examples can include data center collocation services, public cloud, or private cloud locations. </p><p>For on-premises systems, local endpoints within the same location typically connect directly over the <a href="/wiki/LAN" class="mw-redirect" title="LAN">LAN</a>. For remote and external endpoints, available connectivity options mirror those of Hosted or Cloud VoIP solutions. </p><p>However, VoIP traffic to and from the on-premises systems can often also be sent over secure private links. Examples include personal VPN, <a href="/wiki/VPN#Types" class="mw-redirect" title="VPN">site-to-site VPN</a>, private networks such as MPLS and SD-WAN, or via private SBCs (Session Border Controllers). While exceptions and private peering options do exist, it is generally uncommon for those private connectivity methods to be provided by Hosted or Cloud VoIP providers. </p> <div class="mw-heading mw-heading2"><h2 id="Quality_of_service">Quality of service</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=11" title="Edit section: Quality of service"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Communication on the IP network is perceived as less reliable in contrast to the circuit-switched public telephone network because it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental <a href="/wiki/Quality_of_service" title="Quality of service">quality of service</a> (QoS) guarantees. Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to data loss in the presence of congestion<sup id="cite_ref-26" class="reference"><a href="#cite_note-26"><span class="cite-bracket">[</span>b<span class="cite-bracket">]</span></a></sup> than traditional <a href="/wiki/Circuit_switched" class="mw-redirect" title="Circuit switched">circuit switched</a> systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.<sup id="cite_ref-cisco_27-0" class="reference"><a href="#cite_note-cisco-27"><span class="cite-bracket">[</span>25<span class="cite-bracket">]</span></a></sup> Therefore, VoIP implementations may face problems with <a href="/wiki/Network_latency" class="mw-redirect" title="Network latency">latency</a>, packet loss, and <a href="/wiki/Jitter" title="Jitter">jitter</a>.<sup id="cite_ref-cisco_27-1" class="reference"><a href="#cite_note-cisco-27"><span class="cite-bracket">[</span>25<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-28" class="reference"><a href="#cite_note-28"><span class="cite-bracket">[</span>26<span class="cite-bracket">]</span></a></sup> </p><p>By default, network routers handle traffic on a first-come, first-served basis. Fixed delays cannot be controlled as they are caused by the physical distance the packets travel. They are especially problematic when satellite circuits are involved because of the long distance to a <a href="/wiki/Geosynchronous_satellite" title="Geosynchronous satellite">geostationary satellite</a> and back; delays of 400–600 ms are typical. Latency can be minimized by marking voice packets as being delay-sensitive with QoS methods such as <a href="/wiki/DiffServ" class="mw-redirect" title="DiffServ">DiffServ</a>.<sup id="cite_ref-cisco_27-2" class="reference"><a href="#cite_note-cisco-27"><span class="cite-bracket">[</span>25<span class="cite-bracket">]</span></a></sup> </p><p>Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Excessive load on a link can cause congestion and associated <a href="/wiki/Queueing_delay" class="mw-redirect" title="Queueing delay">queueing delays</a> and <a href="/wiki/Packet_loss" title="Packet loss">packet loss</a>. This signals a transport protocol like <a href="/wiki/Transmission_Control_Protocol" title="Transmission Control Protocol">TCP</a> to reduce its transmission rate to alleviate the congestion. But VoIP usually uses <a href="/wiki/User_Datagram_Protocol" title="User Datagram Protocol">UDP</a> not TCP because recovering from congestion through retransmission usually entails too much latency.<sup id="cite_ref-cisco_27-3" class="reference"><a href="#cite_note-cisco-27"><span class="cite-bracket">[</span>25<span class="cite-bracket">]</span></a></sup> So QoS mechanisms can avoid the undesirable loss of VoIP packets by immediately transmitting them ahead of any queued bulk traffic on the same link, even when the link is congested by bulk traffic. </p><p>VoIP endpoints usually have to wait for the completion of transmission of previous packets before new data may be sent. Although it is possible to preempt (abort) a less important packet in mid-transmission, this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets.<sup id="cite_ref-ciscoPacket_29-0" class="reference"><a href="#cite_note-ciscoPacket-29"><span class="cite-bracket">[</span>27<span class="cite-bracket">]</span></a></sup> An alternative to preemption on slower links, such as dialup and <a href="/wiki/Digital_subscriber_line" title="Digital subscriber line">digital subscriber line</a> (DSL), is to reduce the maximum transmission time by reducing the <a href="/wiki/Maximum_transmission_unit" title="Maximum transmission unit">maximum transmission unit</a>. But since every packet must contain protocol headers, this increases relative header overhead on every link traversed.<sup id="cite_ref-ciscoPacket_29-1" class="reference"><a href="#cite_note-ciscoPacket-29"><span class="cite-bracket">[</span>27<span class="cite-bracket">]</span></a></sup> </p><p>The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. <a href="/wiki/Packet_delay_variation" title="Packet delay variation">Packet delay variation</a> results from changes in <a href="/wiki/Queuing_delay" title="Queuing delay">queuing delay</a> along a given network path due to competition from other users for the same transmission links. VoIP receivers accommodate this variation by storing incoming packets briefly in a <a href="/wiki/Playout_buffer" class="mw-redirect" title="Playout buffer">playout buffer</a>, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the <a href="/wiki/Voice_engine" title="Voice engine">voice engine</a> to play it. The added delay is thus a compromise between excessive latency and excessive <a href="/wiki/Dropout_(communications)" title="Dropout (communications)">dropout</a>, i.e. momentary audio interruptions. </p><p>Although jitter is a random variable, it is the sum of several other random variables that are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. Motivated by the <a href="/wiki/Central_limit_theorem" title="Central limit theorem">central limit theorem</a>, jitter can be modeled as a <a href="/wiki/Gaussian_random_variable" class="mw-redirect" title="Gaussian random variable">Gaussian random variable</a>. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful. In practice, the variance in latency of many Internet paths is dominated by a small number (often one) of relatively slow and congested <a href="/wiki/Internet_bottleneck" title="Internet bottleneck">bottleneck links</a>. Most Internet backbone links are now so fast (e.g. 10 Gbit/s) that their delays are dominated by the <a href="/wiki/Signal_transmission" title="Signal transmission">transmission</a> medium (e.g. optical fiber) and the routers driving them do not have enough buffering for queuing delays to be significant.<sup id="cite_ref-30" class="reference"><a href="#cite_note-30"><span class="cite-bracket">[</span>28<span class="cite-bracket">]</span></a></sup> </p><p>A number of protocols have been defined to support the reporting of <a href="/wiki/Quality_of_service" title="Quality of service">quality of service</a> (QoS) and <a href="/wiki/Quality_of_experience" title="Quality of experience">quality of experience</a> (QoE) for VoIP calls. These include <a href="/wiki/RTP_Control_Protocol" title="RTP Control Protocol">RTP Control Protocol</a> (RTCP) extended reports,<sup id="cite_ref-:1_31-0" class="reference"><a href="#cite_note-:1-31"><span class="cite-bracket">[</span>29<span class="cite-bracket">]</span></a></sup> SIP RTCP summary reports, H.460.9 Annex B (for <a href="/wiki/H.323" title="H.323">H.323</a>), <a href="/wiki/H.248" title="H.248">H.248</a>.30 and MGCP extensions. </p><p>The RTCP extended report VoIP metrics block specified by <style data-mw-deduplicate="TemplateStyles:r1238218222">.mw-parser-output cite.citation{font-style:inherit;word-wrap:break-word}.mw-parser-output .citation q{quotes:"\"""\"""'""'"}.mw-parser-output .citation:target{background-color:rgba(0,127,255,0.133)}.mw-parser-output .id-lock-free.id-lock-free a{background:url("//upload.wikimedia.org/wikipedia/commons/6/65/Lock-green.svg")right 0.1em center/9px no-repeat}.mw-parser-output .id-lock-limited.id-lock-limited a,.mw-parser-output .id-lock-registration.id-lock-registration a{background:url("//upload.wikimedia.org/wikipedia/commons/d/d6/Lock-gray-alt-2.svg")right 0.1em center/9px no-repeat}.mw-parser-output .id-lock-subscription.id-lock-subscription a{background:url("//upload.wikimedia.org/wikipedia/commons/a/aa/Lock-red-alt-2.svg")right 0.1em center/9px no-repeat}.mw-parser-output .cs1-ws-icon a{background:url("//upload.wikimedia.org/wikipedia/commons/4/4c/Wikisource-logo.svg")right 0.1em center/12px no-repeat}body:not(.skin-timeless):not(.skin-minerva) .mw-parser-output .id-lock-free a,body:not(.skin-timeless):not(.skin-minerva) .mw-parser-output .id-lock-limited a,body:not(.skin-timeless):not(.skin-minerva) .mw-parser-output .id-lock-registration a,body:not(.skin-timeless):not(.skin-minerva) .mw-parser-output .id-lock-subscription a,body:not(.skin-timeless):not(.skin-minerva) .mw-parser-output .cs1-ws-icon a{background-size:contain;padding:0 1em 0 0}.mw-parser-output .cs1-code{color:inherit;background:inherit;border:none;padding:inherit}.mw-parser-output .cs1-hidden-error{display:none;color:var(--color-error,#d33)}.mw-parser-output .cs1-visible-error{color:var(--color-error,#d33)}.mw-parser-output .cs1-maint{display:none;color:#085;margin-left:0.3em}.mw-parser-output .cs1-kern-left{padding-left:0.2em}.mw-parser-output .cs1-kern-right{padding-right:0.2em}.mw-parser-output .citation .mw-selflink{font-weight:inherit}@media screen{.mw-parser-output .cs1-format{font-size:95%}html.skin-theme-clientpref-night .mw-parser-output .cs1-maint{color:#18911f}}@media screen and (prefers-color-scheme:dark){html.skin-theme-clientpref-os .mw-parser-output .cs1-maint{color:#18911f}}</style><a href="/wiki/RFC_(identifier)" class="mw-redirect" title="RFC (identifier)">RFC</a> <a rel="nofollow" class="external text" href="https://datatracker.ietf.org/doc/html/rfc3611">3611</a> is generated by an VoIP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal/noise/echo level, <a href="/wiki/Mean_opinion_score" title="Mean opinion score">mean opinion scores</a> (MOS) and R factors and configuration information related to the jitter buffer. VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP summary report or one of the other signaling protocol extensions. VoIP metrics reports are intended to support real-time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications. </p> <div class="mw-heading mw-heading3"><h3 id="DSL_and_ATM">DSL and ATM</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=12" title="Edit section: DSL and ATM"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>DSL modems typically provide Ethernet connections to local equipment, but inside they may actually be <a href="/wiki/Asynchronous_Transfer_Mode" title="Asynchronous Transfer Mode">Asynchronous Transfer Mode</a> (ATM) modems.<sup id="cite_ref-32" class="reference"><a href="#cite_note-32"><span class="cite-bracket">[</span>c<span class="cite-bracket">]</span></a></sup> They use <a href="/wiki/ATM_Adaptation_Layer_5" title="ATM Adaptation Layer 5">ATM Adaptation Layer 5</a> (AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission, reassembling them back into Ethernet frames at the receiving end. </p><p>Using a separate <a href="/wiki/Virtual_circuit_identifier" class="mw-redirect" title="Virtual circuit identifier">virtual circuit identifier</a> (VCI) for voice over IP has the potential to reduce latency on shared connections. ATM's potential for latency reduction is greatest on slow links because worst-case latency decreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link, this latency is probably small enough to ensure good VoIP performance without MTU reductions or multiple ATM VCs. The latest generations of DSL, <a href="/wiki/VDSL" title="VDSL">VDSL</a> and <a href="/wiki/VDSL2" class="mw-redirect" title="VDSL2">VDSL2</a>, carry Ethernet without intermediate ATM/AAL5 layers, and they generally support <a href="/wiki/IEEE_802.1p" class="mw-redirect" title="IEEE 802.1p">IEEE 802.1p</a> priority tagging so that VoIP can be queued ahead of less time-critical traffic.<sup id="cite_ref-cisco_27-4" class="reference"><a href="#cite_note-cisco-27"><span class="cite-bracket">[</span>25<span class="cite-bracket">]</span></a></sup> </p><p>ATM has substantial header overhead: 5/53 = 9.4%, roughly twice the total header overhead of a 1500 byte Ethernet frame. This "ATM tax" is incurred by every DSL user whether or not they take advantage of multiple virtual circuits – and few can.<sup id="cite_ref-cisco_27-5" class="reference"><a href="#cite_note-cisco-27"><span class="cite-bracket">[</span>25<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading3"><h3 id="Layer_2">Layer 2</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=13" title="Edit section: Layer 2"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Several protocols are used in the <a href="/wiki/Data_link_layer" title="Data link layer">data link layer</a> and <a href="/wiki/Physical_layer" title="Physical layer">physical layer</a> for quality-of-service mechanisms that help VoIP applications work well even in the presence of <a href="/wiki/Network_congestion" title="Network congestion">network congestion</a>. Some examples include: </p> <ul><li><a href="/wiki/IEEE_802.11e" class="mw-redirect" title="IEEE 802.11e">IEEE 802.11e</a> is an approved amendment to the <a href="/wiki/IEEE_802.11" title="IEEE 802.11">IEEE 802.11</a> standard that defines a set of quality-of-service enhancements for wireless LAN applications through modifications to the <a href="/wiki/Media_access_control" class="mw-redirect" title="Media access control">media access control</a> (MAC) layer. The standard is considered of critical importance for delay-sensitive applications, such as voice over wireless IP.</li> <li><a href="/wiki/IEEE_802.1p" class="mw-redirect" title="IEEE 802.1p">IEEE 802.1p</a> defines 8 different classes of service (including one dedicated to voice) for traffic on layer-2 wired <a href="/wiki/Ethernet" title="Ethernet">Ethernet</a>.</li> <li>The <a href="/wiki/ITU-T" title="ITU-T">ITU-T</a> <a href="/wiki/G.hn" title="G.hn">G.hn</a> standard, which provides a way to create a high-speed (up to 1 gigabit per second) <a href="/wiki/Local_area_network" title="Local area network">Local area network</a> (LAN) using existing home wiring (<a href="/wiki/Power_line_communication" class="mw-redirect" title="Power line communication">power lines</a>, phone lines and <a href="/wiki/Ethernet_over_coax" title="Ethernet over coax">coaxial cables</a>). G.hn provides QoS by means of Contention-Free Transmission Opportunities (CFTXOPs) which are allocated to flows (such as a VoIP call) that require QoS and which have negotiated a <i>contract</i> with the network controllers.</li></ul> <div class="mw-heading mw-heading2"><h2 id="Performance_metrics">Performance metrics</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=14" title="Edit section: Performance metrics"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>The quality of voice transmission is characterized by several metrics that may be monitored by network elements and by the user agent hardware or software. Such metrics include network <a href="/wiki/Packet_loss" title="Packet loss">packet loss</a>, packet <a href="/wiki/Jitter" title="Jitter">jitter</a>, packet <a href="/wiki/Round-trip_delay_time" class="mw-redirect" title="Round-trip delay time">latency</a> (delay), post-dial delay, and echo. The metrics are determined by VoIP performance testing and monitoring.<sup id="cite_ref-33" class="reference"><a href="#cite_note-33"><span class="cite-bracket">[</span>30<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-34" class="reference"><a href="#cite_note-34"><span class="cite-bracket">[</span>31<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-35" class="reference"><a href="#cite_note-35"><span class="cite-bracket">[</span>32<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-36" class="reference"><a href="#cite_note-36"><span class="cite-bracket">[</span>33<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-37" class="reference"><a href="#cite_note-37"><span class="cite-bracket">[</span>34<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-38" class="reference"><a href="#cite_note-38"><span class="cite-bracket">[</span>35<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading2"><h2 id="PSTN_integration">PSTN integration</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=15" title="Edit section: PSTN integration"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <style data-mw-deduplicate="TemplateStyles:r1251242444">.mw-parser-output .ambox{border:1px solid #a2a9b1;border-left:10px solid #36c;background-color:#fbfbfb;box-sizing:border-box}.mw-parser-output .ambox+link+.ambox,.mw-parser-output .ambox+link+style+.ambox,.mw-parser-output .ambox+link+link+.ambox,.mw-parser-output .ambox+.mw-empty-elt+link+.ambox,.mw-parser-output .ambox+.mw-empty-elt+link+style+.ambox,.mw-parser-output .ambox+.mw-empty-elt+link+link+.ambox{margin-top:-1px}html body.mediawiki .mw-parser-output .ambox.mbox-small-left{margin:4px 1em 4px 0;overflow:hidden;width:238px;border-collapse:collapse;font-size:88%;line-height:1.25em}.mw-parser-output .ambox-speedy{border-left:10px solid #b32424;background-color:#fee7e6}.mw-parser-output .ambox-delete{border-left:10px solid #b32424}.mw-parser-output .ambox-content{border-left:10px solid #f28500}.mw-parser-output .ambox-style{border-left:10px solid #fc3}.mw-parser-output .ambox-move{border-left:10px solid #9932cc}.mw-parser-output .ambox-protection{border-left:10px solid #a2a9b1}.mw-parser-output .ambox .mbox-text{border:none;padding:0.25em 0.5em;width:100%}.mw-parser-output .ambox .mbox-image{border:none;padding:2px 0 2px 0.5em;text-align:center}.mw-parser-output .ambox .mbox-imageright{border:none;padding:2px 0.5em 2px 0;text-align:center}.mw-parser-output .ambox .mbox-empty-cell{border:none;padding:0;width:1px}.mw-parser-output .ambox .mbox-image-div{width:52px}@media(min-width:720px){.mw-parser-output .ambox{margin:0 10%}}@media print{body.ns-0 .mw-parser-output .ambox{display:none!important}}</style><table class="box-More_citations_needed_section plainlinks metadata ambox ambox-content ambox-Refimprove" role="presentation"><tbody><tr><td class="mbox-image"><div class="mbox-image-div"><span typeof="mw:File"><a href="/wiki/File:Question_book-new.svg" class="mw-file-description"><img alt="" src="//upload.wikimedia.org/wikipedia/en/thumb/9/99/Question_book-new.svg/50px-Question_book-new.svg.png" decoding="async" width="50" height="39" class="mw-file-element" srcset="//upload.wikimedia.org/wikipedia/en/thumb/9/99/Question_book-new.svg/75px-Question_book-new.svg.png 1.5x, //upload.wikimedia.org/wikipedia/en/thumb/9/99/Question_book-new.svg/100px-Question_book-new.svg.png 2x" data-file-width="512" data-file-height="399" /></a></span></div></td><td class="mbox-text"><div class="mbox-text-span">This section <b>needs additional citations for <a href="/wiki/Wikipedia:Verifiability" title="Wikipedia:Verifiability">verification</a></b>.<span class="hide-when-compact"> Please help <a href="/wiki/Special:EditPage/Voice_over_IP" title="Special:EditPage/Voice over IP">improve this article</a> by <a href="/wiki/Help:Referencing_for_beginners" title="Help:Referencing for beginners">adding citations to reliable sources</a> in this section. Unsourced material may be challenged and removed.</span> <span class="date-container"><i>(<span class="date">November 2019</span>)</i></span><span class="hide-when-compact"><i> (<small><a href="/wiki/Help:Maintenance_template_removal" title="Help:Maintenance template removal">Learn how and when to remove this message</a></small>)</i></span></div></td></tr></tbody></table> <p>A VoIP media gateway controller (aka <a href="/wiki/Class-5_telephone_switch" title="Class-5 telephone switch">Class 5</a> Softswitch) works in cooperation with a media gateway (aka IP Business Gateway) and connects the digital media stream, so as to complete the path for voice and data. Gateways include interfaces for connecting to standard PSTN networks. Ethernet interfaces are also included in the modern systems which are specially designed to link calls that are passed via VoIP.<sup id="cite_ref-39" class="reference"><a href="#cite_note-39"><span class="cite-bracket">[</span>36<span class="cite-bracket">]</span></a></sup> </p><p><a href="/wiki/E.164" title="E.164">E.164</a> is a global numbering standard for both the PSTN and <a href="/wiki/Public_land_mobile_network" title="Public land mobile network">public land mobile network</a> (PLMN). Most VoIP implementations support <a href="/wiki/E.164" title="E.164">E.164</a> to allow calls to be routed to and from VoIP subscribers and the PSTN/PLMN.<sup id="cite_ref-40" class="reference"><a href="#cite_note-40"><span class="cite-bracket">[</span>37<span class="cite-bracket">]</span></a></sup> VoIP implementations can also allow other identification techniques to be used. For example, <a href="/wiki/Skype_Technologies" title="Skype Technologies">Skype</a> allows subscribers to choose <i>Skype names</i> (usernames)<sup id="cite_ref-41" class="reference"><a href="#cite_note-41"><span class="cite-bracket">[</span>38<span class="cite-bracket">]</span></a></sup> whereas SIP implementations can use <a href="/wiki/Uniform_Resource_Identifier" title="Uniform Resource Identifier">Uniform Resource Identifier</a> (URIs) similar to <a href="/wiki/Email_address" title="Email address">email addresses</a>.<sup id="cite_ref-42" class="reference"><a href="#cite_note-42"><span class="cite-bracket">[</span>39<span class="cite-bracket">]</span></a></sup> Often VoIP implementations employ methods of translating non-E.164 identifiers to E.164 numbers and vice versa, such as the Skype-In service provided by Skype<sup id="cite_ref-43" class="reference"><a href="#cite_note-43"><span class="cite-bracket">[</span>40<span class="cite-bracket">]</span></a></sup> and the <a href="/wiki/E.164_number_to_URI_mapping" class="mw-redirect" title="E.164 number to URI mapping">E.164 number to URI mapping</a> (ENUM) service in IMS and SIP.<sup id="cite_ref-44" class="reference"><a href="#cite_note-44"><span class="cite-bracket">[</span>41<span class="cite-bracket">]</span></a></sup> </p><p>Echo can also be an issue for PSTN integration.<sup id="cite_ref-45" class="reference"><a href="#cite_note-45"><span class="cite-bracket">[</span>42<span class="cite-bracket">]</span></a></sup> Common causes of echo include <a href="/wiki/Impedance_matching" title="Impedance matching">impedance mismatches</a> in analog circuitry and an acoustic path from the receive to transmit signal at the receiving end. </p> <div class="mw-heading mw-heading3"><h3 id="Number_portability">Number portability</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=16" title="Edit section: Number portability"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p><a href="/wiki/Local_number_portability" title="Local number portability">Local number portability</a> (LNP) and <a href="/wiki/Mobile_number_portability" title="Mobile number portability">mobile number portability</a> (MNP) also impact VoIP business. Number portability is a service that allows a subscriber to select a new telephone carrier without requiring a new number to be issued. Typically, it is the responsibility of the former carrier to "map" the old number to the undisclosed number assigned by the new carrier. This is achieved by maintaining a database of numbers. A dialed number is initially received by the original carrier and quickly rerouted to the new carrier. Multiple porting references must be maintained even if the subscriber returns to the original carrier. The <a href="/wiki/Federal_Communications_Commission" title="Federal Communications Commission">Federal Communications Commission</a> (FCC) mandates carrier compliance with these consumer-protection stipulations. In November 2007, the FCC in the United States released an order extending number portability obligations to interconnected VoIP providers and carriers that support VoIP providers.<sup id="cite_ref-46" class="reference"><a href="#cite_note-46"><span class="cite-bracket">[</span>43<span class="cite-bracket">]</span></a></sup> </p><p>A voice call originating in the VoIP environment also faces <a href="/wiki/Least-cost_routing" title="Least-cost routing">least-cost routing</a> (LCR) challenges to reach its destination if the number is routed to a mobile phone number on a traditional mobile carrier. LCR is based on checking the destination of each telephone call as it is made, and then sending the call via the network that will cost the customer the least. This rating is subject to some debate given the complexity of call routing created by number portability. With MNP in place, LCR providers can no longer rely on using the network root prefix to determine how to route a call. Instead, they must now determine the actual network of every number before routing the call.<sup id="cite_ref-47" class="reference"><a href="#cite_note-47"><span class="cite-bracket">[</span>44<span class="cite-bracket">]</span></a></sup> </p><p>Therefore, VoIP solutions also need to handle MNP when routing a voice call. In countries without a central database, like the UK, it may be necessary to query the mobile network about which home network a mobile phone number belongs to. As the popularity of VoIP increases in the enterprise markets because of LCR options, VoIP needs to provide a certain level of reliability when handling calls. </p> <div class="mw-heading mw-heading3"><h3 id="Emergency_calls">Emergency calls</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=17" title="Edit section: Emergency calls"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>A telephone connected to a <a href="/wiki/Land_line" class="mw-redirect" title="Land line">land line</a> has a direct relationship between a telephone number and a physical location, which is maintained by the telephone company and available to emergency responders via the national emergency response service centers in form of emergency subscriber lists. When an emergency call is received by a center the location is automatically determined from its databases and displayed on the operator console. </p><p>In IP telephony, no such direct link between location and communications end point exists. Even a provider having wired infrastructure, such as a DSL provider, may know only the approximate location of the device, based on the <a href="/wiki/IP_address" title="IP address">IP address</a> allocated to the network router and the known service address. Some ISPs do not track the automatic assignment of IP addresses to customer equipment.<sup id="cite_ref-fcc_48-0" class="reference"><a href="#cite_note-fcc-48"><span class="cite-bracket">[</span>45<span class="cite-bracket">]</span></a></sup> </p><p>IP communication provides for device mobility. For example, a residential broadband connection may be used as a link to a <a href="/wiki/Virtual_private_network" title="Virtual private network">virtual private network</a> of a corporate entity, in which case the IP address being used for customer communications may belong to the enterprise, not the residential ISP. Such <a href="/wiki/Off-premises_extension" title="Off-premises extension">off-premises extensions</a> may appear as part of an upstream IP PBX. On mobile devices, e.g., a 3G handset or USB wireless broadband adapter, the IP address has no relationship with any physical location known to the telephony service provider, since a mobile user could be anywhere in a region with network coverage, even roaming via another cellular company. </p><p>At the VoIP level, a phone or gateway may identify itself by its account credentials with a <a href="/wiki/Session_Initiation_Protocol" title="Session Initiation Protocol">Session Initiation Protocol</a> (SIP) registrar. In such cases, the <a href="/wiki/Internet_telephony_service_provider" title="Internet telephony service provider">Internet telephony service provider</a> (ITSP) knows only that a particular user's equipment is active. Service providers often provide emergency response services by agreement with the user who registers a physical location and agrees that, if an emergency number is called from the IP device, emergency services are provided to that address only. </p><p>Such emergency services are provided by VoIP vendors in the United States by a system called <a href="/wiki/Enhanced_911" title="Enhanced 911">Enhanced 911</a> (E911), based on the <a href="/wiki/Wireless_Communications_and_Public_Safety_Act" title="Wireless Communications and Public Safety Act">Wireless Communications and Public Safety Act</a>. The VoIP E911 emergency-calling system associates a physical address with the calling party's telephone number. All VoIP providers that provide access to the public switched telephone network are required to implement E911, a service for which the subscriber may be charged. "VoIP providers may not allow customers to opt-out of 911 service."<sup id="cite_ref-fcc_48-1" class="reference"><a href="#cite_note-fcc-48"><span class="cite-bracket">[</span>45<span class="cite-bracket">]</span></a></sup> The <a href="/wiki/VoIP_E911" class="mw-redirect" title="VoIP E911">VoIP E911</a> system is based on a static table lookup. Unlike in cellular phones, where the location of an E911 call can be traced using <a href="/wiki/Assisted_GPS" class="mw-redirect" title="Assisted GPS">assisted GPS</a> or other methods, the VoIP E911 information is accurate only if subscribers keep their emergency address information current.<sup id="cite_ref-49" class="reference"><a href="#cite_note-49"><span class="cite-bracket">[</span>46<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading2"><h2 id="Fax_support">Fax support</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=18" title="Edit section: Fax support"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Sending <a href="/wiki/Fax" title="Fax">faxes</a> over VoIP networks is sometimes referred to as Fax over IP (FoIP). Transmission of fax documents was problematic in early VoIP implementations, as most voice digitization and compression <a href="/wiki/Codec" title="Codec">codecs</a> are optimized for the representation of the human voice and the proper timing of the modem signals cannot be guaranteed in a packet-based, connectionless network. </p><p>A standards-based solution for reliably delivering fax-over-IP is the <a href="/wiki/T.38" title="T.38">T.38</a> protocol. The T.38 protocol is designed to compensate for the differences between traditional packet-less communications over analog lines and packet-based transmissions which are the basis for IP communications. The fax machine may be a standard device connected to an <a href="/wiki/Analog_telephone_adapter" title="Analog telephone adapter">analog telephone adapter</a> (ATA), or it may be a software application or dedicated network device operating via an Ethernet interface.<sup id="cite_ref-50" class="reference"><a href="#cite_note-50"><span class="cite-bracket">[</span>47<span class="cite-bracket">]</span></a></sup> Originally, T.38 was designed to use UDP or TCP transmission methods across an IP network. </p><p>Some newer high-end fax machines have built-in T.38 capabilities which are connected directly to a network switch or router. In T.38 each packet contains a portion of the data stream sent in the previous packet. Two successive packets have to be lost to actually lose <a href="/wiki/Data_integrity" title="Data integrity">data integrity</a>. </p> <div class="mw-heading mw-heading2"><h2 id="Power_requirements">Power requirements</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=19" title="Edit section: Power requirements"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Telephones for traditional residential analog service are usually connected directly to telephone company <a href="/wiki/Phone_line" class="mw-redirect" title="Phone line">phone lines</a> which provide direct current to power most basic analog handsets independently of locally available electrical power. The susceptibility of phone service to power failures is a common problem even with traditional analog service where customers purchase telephone units that operate with wireless handsets to a base station, or that have other modern phone features, such as built-in voicemail or phone book features. </p><p>VoIP phones and VoIP telephone adapters connect to <a href="/wiki/Network_router" class="mw-redirect" title="Network router">routers</a> or <a href="/wiki/Cable_modem" title="Cable modem">cable modems</a> which typically depend on the availability of <a href="/wiki/Mains_electricity" title="Mains electricity">mains electricity</a> or locally generated power.<sup id="cite_ref-51" class="reference"><a href="#cite_note-51"><span class="cite-bracket">[</span>48<span class="cite-bracket">]</span></a></sup> Some VoIP service providers use customer premises equipment (e.g., cable modems) with battery-backed power supplies to assure uninterrupted service for up to several hours in case of local power failures. Such battery-backed devices typically are designed for use with analog handsets. Some VoIP service providers implement services to route calls to other telephone services of the subscriber, such a cellular phone, in the event that the customer's network device is inaccessible to terminate the call. </p> <div class="mw-heading mw-heading2"><h2 id="Security">Security</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=20" title="Edit section: Security"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Secure calls are possible using standardized protocols such as <a href="/wiki/Secure_Real-time_Transport_Protocol" title="Secure Real-time Transport Protocol">Secure Real-time Transport Protocol</a>. Most of the facilities of creating a <a href="/wiki/Secure_telephone" title="Secure telephone">secure telephone</a> connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is necessary only to <a href="/wiki/Encrypt" class="mw-redirect" title="Encrypt">encrypt</a> and <a href="/wiki/Authenticate" class="mw-redirect" title="Authenticate">authenticate</a> the existing data stream. Automated software, such as a <a href="/wiki/Business_telephone_system#Current_trends" title="Business telephone system">virtual PBX</a>, may eliminate the need for personnel to greet and switch incoming calls. </p><p>The security concerns for VoIP telephone systems are similar to those of other Internet-connected devices. This means that <a href="/wiki/Hacker" title="Hacker">hackers</a> with knowledge of <a href="/wiki/VoIP_vulnerabilities" title="VoIP vulnerabilities">VoIP vulnerabilities</a> can perform <a href="/wiki/Denial-of-service" class="mw-redirect" title="Denial-of-service">denial-of-service</a> attacks, harvest customer data, record conversations, and compromise voicemail messages. Compromised VoIP user account or session credentials may enable an attacker to incur substantial charges from third-party services, such as long-distance or international calling. </p><p>The technical details of many VoIP protocols create challenges in routing VoIP traffic through <a href="/wiki/Firewall_(networking)" class="mw-redirect" title="Firewall (networking)">firewalls</a> and <a href="/wiki/Network_address_translator" class="mw-redirect" title="Network address translator">network address translators</a>, used to interconnect to transit networks or the Internet. Private <a href="/wiki/Session_border_controller" title="Session border controller">session border controllers</a> are often employed to enable VoIP calls to and from protected networks. Other methods to <a href="/wiki/NAT_traversal" title="NAT traversal">traverse NAT</a> devices involve assistive protocols such as <a href="/wiki/STUN" title="STUN">STUN</a> and <a href="/wiki/Interactive_Connectivity_Establishment" title="Interactive Connectivity Establishment">Interactive Connectivity Establishment</a> (ICE). </p><p>Standards for securing VoIP are available in the <a href="/wiki/Secure_Real-time_Transport_Protocol" title="Secure Real-time Transport Protocol">Secure Real-time Transport Protocol</a> (SRTP) and the <a href="/wiki/ZRTP" title="ZRTP">ZRTP</a> protocol for <a href="/wiki/Analog_telephony_adapter" class="mw-redirect" title="Analog telephony adapter">analog telephony adapters</a>, as well as for some <a href="/wiki/Softphone" title="Softphone">softphones</a>. <a href="/wiki/IPsec" title="IPsec">IPsec</a> is available to secure <a href="/wiki/Point-to-point_(telecommunications)" title="Point-to-point (telecommunications)">point-to-point</a> VoIP at the transport level by using <a href="/wiki/Opportunistic_encryption" title="Opportunistic encryption">opportunistic encryption</a>. Though many consumer VoIP solutions do not support encryption of the signaling path or the media, securing a VoIP phone is conceptually easier to implement using VoIP than on traditional telephone circuits. A result of the lack of widespread support for encryption is that it is relatively easy to eavesdrop on VoIP calls when access to the data network is possible.<sup id="cite_ref-52" class="reference"><a href="#cite_note-52"><span class="cite-bracket">[</span>49<span class="cite-bracket">]</span></a></sup> Free open-source solutions, such as <a href="/wiki/Wireshark" title="Wireshark">Wireshark</a>, facilitate capturing VoIP conversations. </p><p>Government and military organizations use various security measures to protect VoIP traffic, such as voice over secure IP (VoSIP), secure voice over IP (SVoIP), and secure voice over secure IP (SVoSIP).<sup id="cite_ref-53" class="reference"><a href="#cite_note-53"><span class="cite-bracket">[</span>50<span class="cite-bracket">]</span></a></sup> The distinction lies in whether encryption is applied in the telephone endpoint or in the network.<sup id="cite_ref-54" class="reference"><a href="#cite_note-54"><span class="cite-bracket">[</span>51<span class="cite-bracket">]</span></a></sup> Secure voice over secure IP may be implemented by encrypting the media with protocols such as <a href="/wiki/Secure_Real-time_Transport_Protocol" title="Secure Real-time Transport Protocol">SRTP</a> and <a href="/wiki/ZRTP" title="ZRTP">ZRTP</a>. Secure voice over IP uses <a href="/wiki/Type_1_encryption" class="mw-redirect" title="Type 1 encryption">Type 1 encryption</a> on a classified network, such as <a href="/wiki/SIPRNet" title="SIPRNet">SIPRNet</a>.<sup id="cite_ref-55" class="reference"><a href="#cite_note-55"><span class="cite-bracket">[</span>52<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-56" class="reference"><a href="#cite_note-56"><span class="cite-bracket">[</span>53<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-57" class="reference"><a href="#cite_note-57"><span class="cite-bracket">[</span>54<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-58" class="reference"><a href="#cite_note-58"><span class="cite-bracket">[</span>55<span class="cite-bracket">]</span></a></sup> Public Secure VoIP is also available with free GNU software and in many popular commercial VoIP programs via libraries, such as ZRTP.<sup id="cite_ref-59" class="reference"><a href="#cite_note-59"><span class="cite-bracket">[</span>56<span class="cite-bracket">]</span></a></sup> </p><p>In June 2021, the <a href="/wiki/National_Security_Agency" title="National Security Agency">National Security Agency</a> (NSA) released comprehensive documents describing the four attack planes of a communications system – the network, perimeter, session controllers and <a href="/wiki/Communication_endpoint" title="Communication endpoint">endpoints</a> – and explaining security risks and mitigation techniques for each of them.<sup id="cite_ref-60" class="reference"><a href="#cite_note-60"><span class="cite-bracket">[</span>57<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-61" class="reference"><a href="#cite_note-61"><span class="cite-bracket">[</span>58<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading2"><h2 id="Caller_ID">Caller ID</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=21" title="Edit section: Caller ID"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Voice over IP protocols and equipment provide <a href="/wiki/Caller_ID" title="Caller ID">caller ID</a> support that is compatible with the PSTN. Many VoIP service providers also allow callers to configure custom caller ID information.<sup id="cite_ref-62" class="reference"><a href="#cite_note-62"><span class="cite-bracket">[</span>59<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading2"><h2 id="Hearing_aid_compatibility">Hearing aid compatibility</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=22" title="Edit section: Hearing aid compatibility"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>Wireline telephones which are manufactured in, imported to, or intended to be used in the US with Voice over IP service, on or after February 28, 2020, are required to meet the <a href="/wiki/Hearing_aid" title="Hearing aid">hearing aid</a> compatibility requirements set forth by the <a href="/wiki/Federal_Communications_Commission" title="Federal Communications Commission">Federal Communications Commission</a>.<sup id="cite_ref-63" class="reference"><a href="#cite_note-63"><span class="cite-bracket">[</span>60<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading2"><h2 id="Operational_cost">Operational cost</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=23" title="Edit section: Operational cost"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>VoIP has drastically reduced the cost of communication by sharing network infrastructure between data and voice.<sup id="cite_ref-64" class="reference"><a href="#cite_note-64"><span class="cite-bracket">[</span>61<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-65" class="reference"><a href="#cite_note-65"><span class="cite-bracket">[</span>62<span class="cite-bracket">]</span></a></sup> A single broadband connection has the ability to transmit multiple telephone calls. </p> <div class="mw-heading mw-heading2"><h2 id="Regulatory_and_legal_issues">Regulatory and legal issues</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=24" title="Edit section: Regulatory and legal issues"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1251242444"><table class="box-Update plainlinks metadata ambox ambox-content ambox-Update" role="presentation"><tbody><tr><td class="mbox-image"><div class="mbox-image-div"><span typeof="mw:File"><span><img alt="" src="//upload.wikimedia.org/wikipedia/commons/thumb/b/bd/Ambox_current_red_Asia_Australia.svg/42px-Ambox_current_red_Asia_Australia.svg.png" decoding="async" width="42" height="34" class="mw-file-element" srcset="//upload.wikimedia.org/wikipedia/commons/thumb/b/bd/Ambox_current_red_Asia_Australia.svg/63px-Ambox_current_red_Asia_Australia.svg.png 1.5x, //upload.wikimedia.org/wikipedia/commons/thumb/b/bd/Ambox_current_red_Asia_Australia.svg/84px-Ambox_current_red_Asia_Australia.svg.png 2x" data-file-width="360" data-file-height="290" /></span></span></div></td><td class="mbox-text"><div class="mbox-text-span">This section needs to be <b>updated</b>.<span class="hide-when-compact"> Please help update this article to reflect recent events or newly available information.</span> <span class="date-container"><i>(<span class="date">April 2022</span>)</i></span></div></td></tr></tbody></table> <p>As the popularity of VoIP grows, governments are becoming more interested in regulating VoIP in a manner similar to PSTN services.<sup id="cite_ref-66" class="reference"><a href="#cite_note-66"><span class="cite-bracket">[</span>63<span class="cite-bracket">]</span></a></sup> </p><p>Throughout the developing world, particularly in countries where regulation is weak or <a href="/wiki/Regulatory_capture" title="Regulatory capture">captured</a> by the dominant operator, restrictions on the use of VoIP are often imposed, including in <a href="/wiki/Panama" title="Panama">Panama</a> where VoIP is taxed, Guyana where VoIP is prohibited.<sup id="cite_ref-67" class="reference"><a href="#cite_note-67"><span class="cite-bracket">[</span>64<span class="cite-bracket">]</span></a></sup> In <a href="/wiki/Ethiopia" title="Ethiopia">Ethiopia</a>, where the government is nationalizing telecommunication service, it is a criminal offense to offer services using VoIP. The country has installed firewalls to prevent international calls from being made using VoIP. These measures were taken after the popularity of VoIP reduced the income generated by the state-owned <a href="/wiki/Telecommunications_company" title="Telecommunications company">telecommunications company</a>.<sup class="noprint Inline-Template Template-Fact" style="white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Citation_needed" title="Wikipedia:Citation needed"><span title="This claim needs references to reliable sources. (April 2022)">citation needed</span></a></i>]</sup><sup id="cite_ref-68" class="reference"><a href="#cite_note-68"><span class="cite-bracket">[</span>65<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading3"><h3 id="Canada">Canada</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=25" title="Edit section: Canada"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In <a href="/wiki/Canada" title="Canada">Canada</a>, the <a href="/wiki/Canadian_Radio-television_and_Telecommunications_Commission" title="Canadian Radio-television and Telecommunications Commission">Canadian Radio-television and Telecommunications Commission</a> regulates telephone service, including VoIP telephony service. VoIP services operating in Canada are required to provide <a href="/wiki/9-1-1" class="mw-redirect" title="9-1-1">9-1-1</a> emergency service.<sup id="cite_ref-69" class="reference"><a href="#cite_note-69"><span class="cite-bracket">[</span>66<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading3"><h3 id="European_Union">European Union</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=26" title="Edit section: European Union"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In the <a href="/wiki/European_Union" title="European Union">European Union</a>, the treatment of VoIP service providers is a decision for each national telecommunications regulator, which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations). A general distinction is usually made between VoIP services that function over managed networks (via broadband connections) and VoIP services that function over unmanaged networks (essentially, the Internet).<sup class="noprint Inline-Template Template-Fact" style="white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Citation_needed" title="Wikipedia:Citation needed"><span title="This claim needs references to reliable sources. (September 2013)">citation needed</span></a></i>]</sup> </p><p>The relevant EU Directive is not clearly drafted concerning obligations that can exist independently of market power (e.g., the obligation to offer access to emergency calls), and it is impossible to say definitively whether VoIP service providers of either type are bound by them.<sup class="noprint Inline-Template Template-Fact" style="white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Citation_needed" title="Wikipedia:Citation needed"><span title="This claim needs references to reliable sources. (September 2013)">citation needed</span></a></i>]</sup><sup id="cite_ref-70" class="reference"><a href="#cite_note-70"><span class="cite-bracket">[</span>67<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading3"><h3 id="Arab_states_of_the_GCC">Arab states of the <a href="/wiki/Gulf_Cooperation_Council" title="Gulf Cooperation Council">GCC</a></h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=27" title="Edit section: Arab states of the GCC"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <div class="mw-heading mw-heading4"><h4 id="Oman">Oman</h4><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=28" title="Edit section: Oman"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In <a href="/wiki/Oman" title="Oman">Oman</a>, it is illegal to provide or use unauthorized VoIP services, to the extent that web sites of unlicensed VoIP providers have been blocked.<sup class="noprint Inline-Template Template-Fact" style="white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Citation_needed" title="Wikipedia:Citation needed"><span title="ref below doesn't mention blocked sites (June 2022)">citation needed</span></a></i>]</sup> Violations may be punished with fines of 50,000 Omani Rial (about 130,317 US dollars), a two-year prison sentence or both. In 2009, police raided 121 Internet cafes throughout the country and arrested 212 people for using or providing VoIP services.<sup id="cite_ref-71" class="reference"><a href="#cite_note-71"><span class="cite-bracket">[</span>68<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading4"><h4 id="Saudi_Arabia">Saudi Arabia</h4><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=29" title="Edit section: Saudi Arabia"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In September 2017, <a href="/wiki/Saudi_Arabia" title="Saudi Arabia">Saudi Arabia</a> lifted the ban on VoIPs, in an attempt to reduce operational costs and spur digital entrepreneurship.<sup id="cite_ref-72" class="reference"><a href="#cite_note-72"><span class="cite-bracket">[</span>69<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-73" class="reference"><a href="#cite_note-73"><span class="cite-bracket">[</span>70<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading4"><h4 id="United_Arab_Emirates">United Arab Emirates</h4><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=30" title="Edit section: United Arab Emirates"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In the <a href="/wiki/United_Arab_Emirates" title="United Arab Emirates">United Arab Emirates</a> (UAE), it is illegal to provide or use unauthorized VoIP services. Web sites of unlicensed VoIP providers have been blocked. Some VoIP services such as <a href="/wiki/Skype" title="Skype">Skype</a> were allowed.<sup id="cite_ref-74" class="reference"><a href="#cite_note-74"><span class="cite-bracket">[</span>71<span class="cite-bracket">]</span></a></sup> In January 2018, internet service providers in UAE blocked all VoIP apps, including Skype, but permitting only 2 government-approved VoIP apps (C’ME and BOTIM).<sup id="cite_ref-75" class="reference"><a href="#cite_note-75"><span class="cite-bracket">[</span>72<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-76" class="reference"><a href="#cite_note-76"><span class="cite-bracket">[</span>73<span class="cite-bracket">]</span></a></sup> In opposition, a petition on <i>Change.org</i> garnered over 5000 signatures, in response to which the website was blocked in UAE.<sup id="cite_ref-77" class="reference"><a href="#cite_note-77"><span class="cite-bracket">[</span>74<span class="cite-bracket">]</span></a></sup> </p><p>On March 24, 2020, the United Arab Emirates loosened restriction on VoIP services earlier prohibited in the country, to ease communication during the <a href="/wiki/COVID-19_pandemic" title="COVID-19 pandemic">COVID-19 pandemic</a>. However, popular instant messaging applications like <a href="/wiki/WhatsApp" title="WhatsApp">WhatsApp</a>, <a href="/wiki/Skype" title="Skype">Skype</a>, and <a href="/wiki/FaceTime" title="FaceTime">FaceTime</a> remained blocked from being used for voice and video calls, constricting residents to use paid services from the country's state-owned telecom providers.<sup id="cite_ref-78" class="reference"><a href="#cite_note-78"><span class="cite-bracket">[</span>75<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading3"><h3 id="India">India</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=31" title="Edit section: India"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In <a href="/wiki/India" title="India">India</a>, it is legal to use VoIP, but it is illegal to have <a href="/wiki/VoIP_gateway" class="mw-redirect" title="VoIP gateway">VoIP gateways</a> inside India.<sup id="cite_ref-TRAI-voip-gateway_79-0" class="reference"><a href="#cite_note-TRAI-voip-gateway-79"><span class="cite-bracket">[</span>76<span class="cite-bracket">]</span></a></sup> This effectively means that people who have PCs can use them to make a VoIP call to other computers but not to a normal phone number. Foreign-based VoIP server services are illegal to use in India.<sup id="cite_ref-TRAI-voip-gateway_79-1" class="reference"><a href="#cite_note-TRAI-voip-gateway-79"><span class="cite-bracket">[</span>76<span class="cite-bracket">]</span></a></sup> </p><p>Internet telephony is permitted to the ISP with restrictions. The following services are permitted:<sup id="cite_ref-80" class="reference"><a href="#cite_note-80"><span class="cite-bracket">[</span>77<span class="cite-bracket">]</span></a></sup> </p> <ol><li>PC to PC; within or outside India</li> <li>PC / a device / Adapter conforming to the standard of any international agencies like- ITU or IETF etc. in India to PSTN/PLMN abroad.</li> <li>Any device / Adapter conforming to standards of International agencies like ITU, IETF etc. connected to ISP node with static IP address to similar device / Adapter; within or outside India.</li> <li>Except whatever is described in <span class="cleanup-needed-content" style="padding-left:0.1em; padding-right:0.1em; color:var(--color-subtle, #54595d); border:1px solid var(--border-color-subtle, #c8ccd1);">condition (ii) above</span><sup class="noprint Inline-Template" style="margin-left:0.1em; white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Please_clarify" title="Wikipedia:Please clarify"><span title="There is no condition (ii). This material appears to be a direct quote of the source which is also unclear. (December 2019)">clarification needed</span></a></i>]</sup>, no other form of Internet Telephony is permitted.</li> <li>In India no Separate Numbering Scheme is provided to the Internet Telephony. Presently the 10 digit Numbering allocation based on E.164 is permitted to the Fixed Telephony, GSM, CDMA wireless service. For Internet Telephony, the numbering scheme shall only conform to IP addressing Scheme of <a href="/wiki/Internet_Assigned_Numbers_Authority" title="Internet Assigned Numbers Authority">Internet Assigned Numbers Authority</a> (IANA). Translation of E.164 number / private number to IP address allotted to any device and vice versa, by ISP to show compliance with IANA numbering scheme is not permitted.</li> <li>The Internet Service Licensee is not permitted to have PSTN/PLMN connectivity. Voice communication to and from a telephone connected to PSTN/PLMN and following E.164 numbering is prohibited in India.</li></ol> <div class="mw-heading mw-heading3"><h3 id="South_Korea">South Korea</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=32" title="Edit section: South Korea"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In <a href="/wiki/South_Korea" title="South Korea">South Korea</a>, only providers registered with the government are authorized to offer VoIP services. Unlike many VoIP providers, most of whom offer flat rates, Korean VoIP services are generally metered and charged at rates similar to terrestrial calling. Foreign VoIP providers encounter high barriers to government registration. This issue came to a head in 2006 when <a href="/wiki/Internet_service_providers" class="mw-redirect" title="Internet service providers">Internet service providers</a> providing personal Internet services by contract to <a href="/wiki/United_States_Forces_Korea" title="United States Forces Korea">United States Forces Korea</a> (USFK) members residing on USFK bases threatened to block off access to VoIP services used by USFK members as an economical way to keep in contact with their families in the United States, on the grounds that the service members' VoIP providers were not registered. A compromise was reached between USFK and Korean telecommunications officials in January 2007, wherein USFK service members arriving in Korea before June 1, 2007, and subscribing to the ISP services provided on base could continue to use their US-based VoIP subscription, but later arrivals are required to use a Korean-based VoIP provider, which by contract will offer pricing similar to the flat rates offered by US VoIP providers.<sup id="cite_ref-81" class="reference"><a href="#cite_note-81"><span class="cite-bracket">[</span>78<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading3"><h3 id="United_States">United States</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=33" title="Edit section: United States"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>In the United States, the FCC requires all interconnected VoIP service providers to comply with requirements comparable to those for traditional telecommunications service providers.<sup id="cite_ref-82" class="reference"><a href="#cite_note-82"><span class="cite-bracket">[</span>79<span class="cite-bracket">]</span></a></sup> VoIP operators in the US are required to support <a href="/wiki/Local_number_portability" title="Local number portability">local number portability</a>; make service accessible to people with disabilities; pay regulatory fees, <a href="/wiki/Universal_service" title="Universal service">universal service</a> contributions, and other mandated payments; and enable law enforcement authorities to conduct surveillance pursuant to the <a href="/wiki/Communications_Assistance_for_Law_Enforcement_Act" title="Communications Assistance for Law Enforcement Act">Communications Assistance for Law Enforcement Act</a> (CALEA). </p><p>Operators of <i>Interconnected</i> VoIP (fully connected to the PSTN) are mandated to provide <a href="/wiki/Enhanced_911" title="Enhanced 911">Enhanced 911</a> service without special request, provide for customer location updates, clearly disclose any limitations on their E-911 functionality to their consumers, obtain affirmative acknowledgements of these disclosures from all consumers,<sup id="cite_ref-83" class="reference"><a href="#cite_note-83"><span class="cite-bracket">[</span>80<span class="cite-bracket">]</span></a></sup> and may not allow their customers to opt-out of 911 service.<sup id="cite_ref-FCC-VoIP-911_84-0" class="reference"><a href="#cite_note-FCC-VoIP-911-84"><span class="cite-bracket">[</span>81<span class="cite-bracket">]</span></a></sup> VoIP operators also receive the benefit of certain US telecommunications regulations, including an entitlement to <a href="/wiki/Interconnection" title="Interconnection">interconnection</a> and exchange of traffic with <a href="/wiki/Incumbent_local_exchange_carrier" title="Incumbent local exchange carrier">incumbent local exchange carriers</a> via wholesale carriers. Providers of <i>nomadic</i> VoIP service—those who are unable to determine the location of their users—are exempt from state telecommunications regulation.<sup id="cite_ref-85" class="reference"><a href="#cite_note-85"><span class="cite-bracket">[</span>82<span class="cite-bracket">]</span></a></sup> </p><p>Another legal issue that the <a href="/wiki/US_Congress" class="mw-redirect" title="US Congress">US Congress</a> is debating concerns changes to the <a href="/wiki/Foreign_Intelligence_Surveillance_Act" title="Foreign Intelligence Surveillance Act">Foreign Intelligence Surveillance Act</a>. The issue in question is calls between Americans and foreigners. The NSA is not authorized to tap Americans' conversations without a warrant—but the Internet, and specifically VoIP does not draw as clear a line to the location of a caller or a call's recipient as the traditional phone system does. As VoIP's low cost and flexibility convinces more and more organizations to adopt the technology, surveillance for law enforcement agencies becomes more difficult. VoIP technology has also increased federal security concerns because VoIP and similar technologies have made it more difficult for the government to determine where a target is physically located when communications are being intercepted, and that creates a whole set of new legal challenges.<sup id="cite_ref-Greenberg_86-0" class="reference"><a href="#cite_note-Greenberg-86"><span class="cite-bracket">[</span>83<span class="cite-bracket">]</span></a></sup> </p> <div class="mw-heading mw-heading2"><h2 id="History">History</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=34" title="Edit section: History"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <p>The early developments of <a href="/wiki/Packet_network" class="mw-redirect" title="Packet network">packet network</a> designs by <a href="/wiki/Paul_Baran" title="Paul Baran">Paul Baran</a> 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 <a href="/wiki/Telecommunications" title="Telecommunications">telecommunications</a> of the mid-twentieth century. <a href="/wiki/Danny_Cohen_(engineer)" class="mw-redirect" title="Danny Cohen (engineer)">Danny Cohen</a> first demonstrated a form of <a href="/wiki/Packet_telephony" title="Packet telephony">packet voice</a> in 1973 which was developed into <a href="/wiki/Network_Voice_Protocol" title="Network Voice Protocol">Network Voice Protocol</a> which operated across the early <a href="/wiki/ARPANET" title="ARPANET">ARPANET</a>.<sup id="cite_ref-87" class="reference"><a href="#cite_note-87"><span class="cite-bracket">[</span>84<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-88" class="reference"><a href="#cite_note-88"><span class="cite-bracket">[</span>85<span class="cite-bracket">]</span></a></sup> </p><p>On the early ARPANET, real-time voice communication was not possible with uncompressed <a href="/wiki/Pulse-code_modulation" title="Pulse-code modulation">pulse-code modulation</a> (PCM) <a href="/wiki/Digital_audio" title="Digital audio">digital speech</a> packets, which had a <a href="/wiki/Bit_rate" title="Bit rate">bit rate</a> of 64<span class="nowrap"> </span>kbps, much greater than the 2.4<span class="nowrap"> </span>kbps <a href="/wiki/Internet_bandwidth" class="mw-redirect" title="Internet bandwidth">bandwidth</a> of early <a href="/wiki/Modems" class="mw-redirect" title="Modems">modems</a>. The solution to this problem was <a href="/wiki/Linear_predictive_coding" title="Linear predictive coding">linear predictive coding</a> (LPC), a <a href="/wiki/Speech_coding" title="Speech coding">speech coding</a> <a href="/wiki/Data_compression" title="Data compression">data compression</a> algorithm that was first proposed by <a href="/wiki/Fumitada_Itakura" title="Fumitada Itakura">Fumitada Itakura</a> of <a href="/wiki/Nagoya_University" title="Nagoya University">Nagoya University</a> and Shuzo Saito of <a href="/wiki/Nippon_Telegraph_and_Telephone" title="Nippon Telegraph and Telephone">Nippon Telegraph and Telephone</a> (NTT) in 1966. LPC was capable of speech compression down to 2.4<span class="nowrap"> </span>kbps, leading to the first successful real-time conversation over ARPANET in 1974, between Culler-Harrison Incorporated in <a href="/wiki/Goleta,_California" title="Goleta, California">Goleta, California</a>, and <a href="/wiki/MIT_Lincoln_Laboratory" title="MIT Lincoln Laboratory">MIT Lincoln Laboratory</a> in <a href="/wiki/Lexington,_Massachusetts" title="Lexington, Massachusetts">Lexington, Massachusetts</a>.<sup id="cite_ref-Gray_89-0" class="reference"><a href="#cite_note-Gray-89"><span class="cite-bracket">[</span>86<span class="cite-bracket">]</span></a></sup> LPC has since been the most widely used speech coding method.<sup id="cite_ref-90" class="reference"><a href="#cite_note-90"><span class="cite-bracket">[</span>87<span class="cite-bracket">]</span></a></sup> <a href="/wiki/Code-excited_linear_prediction" title="Code-excited linear prediction">Code-excited linear prediction</a> (CELP), a type of LPC algorithm, was developed by <a href="/wiki/Manfred_R._Schroeder" title="Manfred R. Schroeder">Manfred R. Schroeder</a> and <a href="/wiki/Bishnu_S._Atal" title="Bishnu S. Atal">Bishnu S. Atal</a> in 1985.<sup id="cite_ref-Schroeder_91-0" class="reference"><a href="#cite_note-Schroeder-91"><span class="cite-bracket">[</span>88<span class="cite-bracket">]</span></a></sup> LPC algorithms remain an <a href="/wiki/Audio_coding_standard" class="mw-redirect" title="Audio coding standard">audio coding standard</a> in modern VoIP technology.<sup id="cite_ref-Gray_89-1" class="reference"><a href="#cite_note-Gray-89"><span class="cite-bracket">[</span>86<span class="cite-bracket">]</span></a></sup> </p><p>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 <a href="/wiki/Internet" title="Internet">Internet</a> 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 <a href="/wiki/VocalTec" title="VocalTec">VocalTec</a>, based on a patent by <a href="/wiki/Lior_Haramaty" title="Lior Haramaty">Lior Haramaty</a> and <a href="/wiki/Alon_Cohen" title="Alon Cohen">Alon Cohen</a>,<sup id="cite_ref-92" class="reference"><a href="#cite_note-92"><span class="cite-bracket">[</span>89<span class="cite-bracket">]</span></a></sup> 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 <a href="/wiki/Marian_Croak" title="Marian Croak">Marian Croak</a> and her team filed many patents related to the technology.<sup class="noprint Inline-Template Template-Fact" style="white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Citation_needed" title="Wikipedia:Citation needed"><span title="Marian Croak and sources there are not clear on exactly when this work occurred. (March 2024)">citation needed</span></a></i>]</sup> By the late 1990s, the first <a href="/wiki/Softswitch" title="Softswitch">softswitches</a> became available, and new protocols, such as <a href="/wiki/H.323" title="H.323">H.323</a>, MGCP and <a href="/wiki/Session_Initiation_Protocol" title="Session Initiation Protocol">Session Initiation Protocol</a> (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 <a href="/wiki/Asterisk_PBX" class="mw-redirect" title="Asterisk PBX">Asterisk PBX</a>, fueled widespread interest and entrepreneurship in voice-over-IP services, applying new Internet technology paradigms, such as <a href="/wiki/Cloud_service" class="mw-redirect" title="Cloud service">cloud services</a> to telephony. </p> <div class="mw-heading mw-heading3"><h3 id="Milestones">Milestones</h3><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=35" title="Edit section: Milestones"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <ul><li>1966: <a href="/wiki/Linear_predictive_coding" title="Linear predictive coding">Linear predictive coding</a> (LPC) proposed by <a href="/wiki/Fumitada_Itakura" title="Fumitada Itakura">Fumitada Itakura</a> of <a href="/wiki/Nagoya_University" title="Nagoya University">Nagoya University</a> and Shuzo Saito of <a href="/wiki/Nippon_Telegraph_and_Telephone" title="Nippon Telegraph and Telephone">Nippon Telegraph and Telephone</a> (NTT).<sup id="cite_ref-Gray_89-2" class="reference"><a href="#cite_note-Gray-89"><span class="cite-bracket">[</span>86<span class="cite-bracket">]</span></a></sup></li> <li>1973: <a href="/wiki/Packet_telephony" title="Packet telephony">Packet voice</a> application by <a href="/wiki/Danny_Cohen_(engineer)" class="mw-redirect" title="Danny Cohen (engineer)">Danny Cohen</a>.</li> <li>1974: The <a href="/wiki/Institute_of_Electrical_and_Electronics_Engineers" title="Institute of Electrical and Electronics Engineers">Institute of Electrical and Electronics Engineers</a> (IEEE) publishes a paper entitled "A Protocol for Packet Network Interconnection".<sup id="cite_ref-93" class="reference"><a href="#cite_note-93"><span class="cite-bracket">[</span>90<span class="cite-bracket">]</span></a></sup></li> <li>1974: <a href="/wiki/Network_Voice_Protocol" title="Network Voice Protocol">Network Voice Protocol</a> (NVP) tested over <a href="/wiki/ARPANET" title="ARPANET">ARPANET</a> in August 1974, carrying barely intelligible 16<span class="nowrap"> </span>kpbs <a href="/wiki/CVSD" class="mw-redirect" title="CVSD">CVSD</a> encoded voice.<sup id="cite_ref-Gray_89-3" class="reference"><a href="#cite_note-Gray-89"><span class="cite-bracket">[</span>86<span class="cite-bracket">]</span></a></sup></li> <li>1974: The first successful real-time conversation over ARPANET achieved using 2.4<span class="nowrap"> </span>kpbs LPC, between Culler-Harrison Incorporated in <a href="/wiki/Goleta,_California" title="Goleta, California">Goleta, California</a>, and <a href="/wiki/MIT_Lincoln_Laboratory" title="MIT Lincoln Laboratory">MIT Lincoln Laboratory</a> in <a href="/wiki/Lexington,_Massachusetts" title="Lexington, Massachusetts">Lexington, Massachusetts</a>.<sup id="cite_ref-Gray_89-4" class="reference"><a href="#cite_note-Gray-89"><span class="cite-bracket">[</span>86<span class="cite-bracket">]</span></a></sup></li> <li>1977: Danny Cohen and <a href="/wiki/Jon_Postel" title="Jon Postel">Jon Postel</a> of the USC <a href="/wiki/Information_Sciences_Institute" title="Information Sciences Institute">Information Sciences Institute</a>, and <a href="/wiki/Vint_Cerf" title="Vint Cerf">Vint Cerf</a> of the Defense Advanced Research Projects Agency (DARPA), agree to separate IP from TCP, and create UDP for carrying real-time traffic.</li> <li>1981: <a href="/wiki/IPv4" title="IPv4">IPv4</a> is described in RFC 791.</li> <li>1985: The <a href="/wiki/National_Science_Foundation" title="National Science Foundation">National Science Foundation</a> commissions the creation of <a href="/wiki/NSFNET" class="mw-redirect" title="NSFNET">NSFNET</a>.<sup id="cite_ref-94" class="reference"><a href="#cite_note-94"><span class="cite-bracket">[</span>91<span class="cite-bracket">]</span></a></sup></li> <li>1985: <a href="/wiki/Code-excited_linear_prediction" title="Code-excited linear prediction">Code-excited linear prediction</a> (CELP), a type of LPC algorithm, developed by <a href="/wiki/Manfred_R._Schroeder" title="Manfred R. Schroeder">Manfred R. Schroeder</a> and <a href="/wiki/Bishnu_S._Atal" title="Bishnu S. Atal">Bishnu S. Atal</a>.<sup id="cite_ref-Schroeder_91-1" class="reference"><a href="#cite_note-Schroeder-91"><span class="cite-bracket">[</span>88<span class="cite-bracket">]</span></a></sup></li> <li>1986: Proposals from various standards organizations<sup class="noprint Inline-Template" style="white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Citing_sources" title="Wikipedia:Citing sources"><span title="Statement needs to be more specific about the content to which it refers. (April 2012)">specify</span></a></i>]</sup> for <a href="/wiki/VoATM" title="VoATM">Voice over ATM</a>, in addition to commercial packet voice products from companies such as <a href="/wiki/StrataCom" title="StrataCom">StrataCom</a></li> <li>1991: Speak Freely, a voice-over-IP application, was released to the public domain.<sup id="cite_ref-B2C_95-0" class="reference"><a href="#cite_note-B2C-95"><span class="cite-bracket">[</span>92<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-McCraw_96-0" class="reference"><a href="#cite_note-McCraw-96"><span class="cite-bracket">[</span>93<span class="cite-bracket">]</span></a></sup></li> <li>1992: The Frame Relay Forum conducts development of standards for voice over <a href="/wiki/Frame_Relay" title="Frame Relay">Frame Relay</a>.</li> <li>1992: <a href="/wiki/InSoft_Inc." title="InSoft Inc.">InSoft Inc.</a> announces and launches its desktop conferencing product Communique, which includes VoIP and video.<sup id="cite_ref-B2C_95-1" class="reference"><a href="#cite_note-B2C-95"><span class="cite-bracket">[</span>92<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-Inc1992_97-0" class="reference"><a href="#cite_note-Inc1992-97"><span class="cite-bracket">[</span>94<span class="cite-bracket">]</span></a></sup> 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.<sup class="noprint Inline-Template Template-Fact" style="white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Citation_needed" title="Wikipedia:Citation needed"><span title="This claim needs references to reliable sources. (July 2023)">citation needed</span></a></i>]</sup></li> <li>1993 Release of VocalChat, a commercial packet network PC voice communication software from <a href="/wiki/VocalTec" title="VocalTec">VocalTec</a>.<sup class="noprint Inline-Template Template-Fact" style="white-space:nowrap;">[<i><a href="/wiki/Wikipedia:Citation_needed" title="Wikipedia:Citation needed"><span title="no occurrence of VocalChat at VocalTec (July 2023)">citation needed</span></a></i>]</sup></li> <li>1994: MTALK, a freeware LAN VoIP application for <a href="/wiki/Linux" title="Linux">Linux</a><sup id="cite_ref-98" class="reference"><a href="#cite_note-98"><span class="cite-bracket">[</span>95<span class="cite-bracket">]</span></a></sup></li> <li>1995: <ul><li><a href="/wiki/VocalTec" title="VocalTec">VocalTec</a> releases <i>Internet Phone</i> commercial Internet phone software.<sup id="cite_ref-99" class="reference"><a href="#cite_note-99"><span class="cite-bracket">[</span>96<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-100" class="reference"><a href="#cite_note-100"><span class="cite-bracket">[</span>97<span class="cite-bracket">]</span></a></sup></li> <li><a href="/wiki/Intel" title="Intel">Intel</a>, <a href="/wiki/Microsoft" title="Microsoft">Microsoft</a> and <a href="/wiki/Radvision" title="Radvision">Radvision</a> initiated standardization activities for VoIP communications system.<sup id="cite_ref-101" class="reference"><a href="#cite_note-101"><span class="cite-bracket">[</span>98<span class="cite-bracket">]</span></a></sup></li></ul></li> <li>1996: <ul><li><a href="/wiki/ITU-T" title="ITU-T">ITU-T</a> begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the <a href="/wiki/H.323" title="H.323">H.323</a> standard.<sup id="cite_ref-102" class="reference"><a href="#cite_note-102"><span class="cite-bracket">[</span>99<span class="cite-bracket">]</span></a></sup></li> <li>US telecommunications companies petition the US Congress to ban Internet phone technology.<sup id="cite_ref-103" class="reference"><a href="#cite_note-103"><span class="cite-bracket">[</span>100<span class="cite-bracket">]</span></a></sup></li> <li><a href="/wiki/G.729" title="G.729">G.729</a> speech codec introduced, using CELP (LPC) algorithm.<sup id="cite_ref-104" class="reference"><a href="#cite_note-104"><span class="cite-bracket">[</span>101<span class="cite-bracket">]</span></a></sup></li></ul></li> <li>1997: <a href="/wiki/Level_3_Communications" title="Level 3 Communications">Level 3</a> began development of its first <a href="/wiki/Softswitch" title="Softswitch">softswitch</a>, a term they coined in 1998.<sup id="cite_ref-ilocus_softswitch_105-0" class="reference"><a href="#cite_note-ilocus_softswitch-105"><span class="cite-bracket">[</span>102<span class="cite-bracket">]</span></a></sup></li> <li>1999: <ul><li>The <a href="/wiki/Session_Initiation_Protocol" title="Session Initiation Protocol">Session Initiation Protocol</a> (SIP) specification RFC 2543 is released.<sup id="cite_ref-106" class="reference"><a href="#cite_note-106"><span class="cite-bracket">[</span>103<span class="cite-bracket">]</span></a></sup></li> <li><a href="/wiki/Mark_Spencer_(computer_engineer)" title="Mark Spencer (computer engineer)">Mark Spencer</a> of <a href="/wiki/Digium" title="Digium">Digium</a> develops <a href="/wiki/Asterisk_(PBX)" title="Asterisk (PBX)">Asterisk</a>, the first <a href="/wiki/Open_source_software" class="mw-redirect" title="Open source software">open source</a> <a href="/wiki/Private_branch_exchange" class="mw-redirect" title="Private branch exchange">private branch exchange</a> (PBX) software.<sup id="cite_ref-107" class="reference"><a href="#cite_note-107"><span class="cite-bracket">[</span>104<span class="cite-bracket">]</span></a></sup></li> <li>A <a href="/wiki/Discrete_cosine_transform" title="Discrete cosine transform">discrete cosine transform</a> (DCT) variant called the <a href="/wiki/Modified_discrete_cosine_transform" title="Modified discrete cosine transform">modified discrete cosine transform</a> (MDCT) is adopted for the <a href="/wiki/Siren_(codec)" title="Siren (codec)">Siren</a> codec, used in the <a href="/wiki/G.722.1" title="G.722.1">G.722.1</a> <a href="/wiki/Wideband_audio" title="Wideband audio">wideband audio</a> coding standard.<sup id="cite_ref-Hersent_108-0" class="reference"><a href="#cite_note-Hersent-108"><span class="cite-bracket">[</span>105<span class="cite-bracket">]</span></a></sup><sup id="cite_ref-Lutzky_109-0" class="reference"><a href="#cite_note-Lutzky-109"><span class="cite-bracket">[</span>106<span class="cite-bracket">]</span></a></sup></li> <li>The MDCT is adapted into the LD-MDCT algorithm, used in the <a href="/wiki/AAC-LD" title="AAC-LD">AAC-LD</a> standard.<sup id="cite_ref-Schnell_110-0" class="reference"><a href="#cite_note-Schnell-110"><span class="cite-bracket">[</span>107<span class="cite-bracket">]</span></a></sup></li></ul></li> <li>2001: <a href="/wiki/INOC-DBA" title="INOC-DBA">INOC-DBA</a>, the first inter-provider <a href="/wiki/Session_Initiation_Protocol" title="Session Initiation Protocol">SIP</a> network is deployed; this is also the first voice network to reach all seven continents.<sup id="cite_ref-111" class="reference"><a href="#cite_note-111"><span class="cite-bracket">[</span>108<span class="cite-bracket">]</span></a></sup></li> <li>2003: <a href="/wiki/Skype" title="Skype">Skype</a> 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.</li> <li>2004: Commercial VoIP service providers proliferate.</li> <li>2005: <a href="/wiki/PhoneGnome" title="PhoneGnome">PhoneGnome</a> VoIP service is launched by TelEvolution, Inc. of California.<sup id="cite_ref-112" class="reference"><a href="#cite_note-112"><span class="cite-bracket">[</span>109<span class="cite-bracket">]</span></a></sup></li> <li>2006: <a href="/wiki/G.729.1" title="G.729.1">G.729.1</a> wideband codec introduced, using MDCT and CELP (LPC) algorithms.<sup id="cite_ref-Nagireddi_113-0" class="reference"><a href="#cite_note-Nagireddi-113"><span class="cite-bracket">[</span>110<span class="cite-bracket">]</span></a></sup></li> <li>2007: VoIP device manufacturers and sellers boom in Asia, specifically in the Philippines where many families of overseas workers reside.<sup id="cite_ref-114" class="reference"><a href="#cite_note-114"><span class="cite-bracket">[</span>111<span class="cite-bracket">]</span></a></sup></li> <li>2009: <a href="/wiki/SILK" title="SILK">SILK</a> codec introduced, using LPC algorithm,<sup id="cite_ref-IETF79_115-0" class="reference"><a href="#cite_note-IETF79-115"><span class="cite-bracket">[</span>112<span class="cite-bracket">]</span></a></sup> and used for voice calling in <a href="/wiki/Skype" title="Skype">Skype</a>.<sup id="cite_ref-116" class="reference"><a href="#cite_note-116"><span class="cite-bracket">[</span>113<span class="cite-bracket">]</span></a></sup></li> <li>2010: <a href="/wiki/Apple_Inc." title="Apple Inc.">Apple</a> introduces <a href="/wiki/FaceTime" title="FaceTime">FaceTime</a>, which uses the LD-MDCT-based AAC-LD codec.<sup id="cite_ref-AppleInsider_standards_1_117-0" class="reference"><a href="#cite_note-AppleInsider_standards_1-117"><span class="cite-bracket">[</span>114<span class="cite-bracket">]</span></a></sup></li> <li>2011: <ul><li>Rise of <a href="/wiki/WebRTC" title="WebRTC">WebRTC</a> technology which supports VoIP directly in browsers.</li> <li><a href="/wiki/CELT" title="CELT">CELT</a> codec introduced, using MDCT algorithm.<sup id="cite_ref-presentation_118-0" class="reference"><a href="#cite_note-presentation-118"><span class="cite-bracket">[</span>115<span class="cite-bracket">]</span></a></sup></li></ul></li> <li>2012: <a href="/wiki/Opus_(audio_format)" title="Opus (audio format)">Opus</a> codec introduced, using MDCT and LPC algorithms.<sup id="cite_ref-Valin_119-0" class="reference"><a href="#cite_note-Valin-119"><span class="cite-bracket">[</span>116<span class="cite-bracket">]</span></a></sup></li></ul> <div class="mw-heading mw-heading2"><h2 id="See_also">See also</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=36" title="Edit section: See also"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <style data-mw-deduplicate="TemplateStyles:r1184024115">.mw-parser-output .div-col{margin-top:0.3em;column-width:30em}.mw-parser-output .div-col-small{font-size:90%}.mw-parser-output .div-col-rules{column-rule:1px solid #aaa}.mw-parser-output .div-col dl,.mw-parser-output .div-col ol,.mw-parser-output .div-col ul{margin-top:0}.mw-parser-output .div-col li,.mw-parser-output .div-col dd{page-break-inside:avoid;break-inside:avoid-column}</style><div class="div-col" style="column-width: 20em;"> <ul><li><a href="/wiki/Audio_over_IP" title="Audio over IP">Audio over IP</a></li> <li><a href="/wiki/Call-through_telecom" title="Call-through telecom">Call-through telecom</a></li> <li><a href="/wiki/Comparison_of_audio_network_protocols" title="Comparison of audio network protocols">Comparison of audio network protocols</a></li> <li><a href="/wiki/Comparison_of_VoIP_software" title="Comparison of VoIP software">Comparison of VoIP software</a></li> <li><a href="/wiki/Differentiated_services" title="Differentiated services">Differentiated services</a></li> <li><a href="/wiki/High_Bit_Rate_Media_Transport" class="mw-redirect" title="High Bit Rate Media Transport">High Bit Rate Media Transport</a></li> <li><a href="/wiki/Integrated_services" title="Integrated services">Integrated services</a></li> <li><a href="/wiki/Internet_fax" title="Internet fax">Internet fax</a></li> <li><a href="/wiki/IP_Multimedia_Subsystem" title="IP Multimedia Subsystem">IP Multimedia Subsystem</a></li> <li><a href="/wiki/List_of_VoIP_companies" title="List of VoIP companies">List of VoIP companies</a></li> <li><a href="/wiki/Mobile_VoIP" title="Mobile VoIP">Mobile VoIP</a></li> <li><a href="/wiki/RTP_payload_formats" title="RTP payload formats">RTP payload formats</a></li> <li><a href="/wiki/SIP_trunking" title="SIP trunking">SIP trunking</a></li> <li><a href="/wiki/UNIStim" title="UNIStim">UNIStim</a></li> <li><a href="/wiki/Voice_over_LTE" title="Voice over LTE">Voice over LTE</a></li> <li><a href="/wiki/VoiceXML" title="VoiceXML">VoiceXML</a></li> <li><a href="/wiki/VoIP_VPN" title="VoIP VPN">VoIP VPN</a></li> <li><a href="/wiki/VoIP_recording" title="VoIP recording">VoIP recording</a></li></ul> </div> <div class="mw-heading mw-heading2"><h2 id="Notes">Notes</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=37" title="Edit section: Notes"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <style data-mw-deduplicate="TemplateStyles:r1239543626">.mw-parser-output .reflist{margin-bottom:0.5em;list-style-type:decimal}@media screen{.mw-parser-output .reflist{font-size:90%}}.mw-parser-output .reflist .references{font-size:100%;margin-bottom:0;list-style-type:inherit}.mw-parser-output .reflist-columns-2{column-width:30em}.mw-parser-output .reflist-columns-3{column-width:25em}.mw-parser-output .reflist-columns{margin-top:0.3em}.mw-parser-output .reflist-columns ol{margin-top:0}.mw-parser-output .reflist-columns li{page-break-inside:avoid;break-inside:avoid-column}.mw-parser-output .reflist-upper-alpha{list-style-type:upper-alpha}.mw-parser-output .reflist-upper-roman{list-style-type:upper-roman}.mw-parser-output .reflist-lower-alpha{list-style-type:lower-alpha}.mw-parser-output .reflist-lower-greek{list-style-type:lower-greek}.mw-parser-output .reflist-lower-roman{list-style-type:lower-roman}</style><div class="reflist reflist-lower-alpha"> <div class="mw-references-wrap"><ol class="references"> <li id="cite_note-2"><span class="mw-cite-backlink"><b><a href="#cite_ref-2">^</a></b></span> <span class="reference-text">Variously pronounced as individual letters, <span class="nowrap"><i>V-O-I-P</i>,</span> or as a word, <span class="rt-commentedText nowrap"><span class="IPA nopopups noexcerpt" lang="en-fonipa"><a href="/wiki/Help:IPA/English" title="Help:IPA/English">/<span style="border-bottom:1px dotted"><span title="'v' in 'vie'">v</span><span title="/ɔɪ/: 'oi' in 'choice'">ɔɪ</span><span title="'p' in 'pie'">p</span></span>/</a></span></span> (<a href="/wiki/Help:Pronunciation_respelling_key" title="Help:Pronunciation respelling key"><i title="English pronunciation respelling"><span style="font-size:90%">VOYP</span></i></a>)<sup id="cite_ref-1" class="reference"><a href="#cite_note-1"><span class="cite-bracket">[</span>1<span class="cite-bracket">]</span></a></sup></span> </li> <li id="cite_note-26"><span class="mw-cite-backlink"><b><a href="#cite_ref-26">^</a></b></span> <span class="reference-text">IP networks may also be more prone to <a href="/wiki/DoS_attacks" class="mw-redirect" title="DoS attacks">DoS attacks</a> that cause congestion.<sup id="cite_ref-25" class="reference"><a href="#cite_note-25"><span class="cite-bracket">[</span>24<span class="cite-bracket">]</span></a></sup></span> </li> <li id="cite_note-32"><span class="mw-cite-backlink"><b><a href="#cite_ref-32">^</a></b></span> <span class="reference-text">Technologies such as <a href="/wiki/802.3ah" class="mw-redirect" title="802.3ah">802.3ah</a> can be used for DSL connectivity without using ATM.</span> </li> </ol></div></div> <div class="mw-heading mw-heading2"><h2 id="References">References</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=38" title="Edit section: References"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1239543626"><div class="reflist"> <div class="mw-references-wrap mw-references-columns"><ol class="references"> <li id="cite_note-1"><span class="mw-cite-backlink"><b><a href="#cite_ref-1">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite class="citation encyclopaedia cs1"><a rel="nofollow" class="external text" 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Archived from <a rel="nofollow" class="external text" href="https://www.cse.wustl.edu/~jain/cis788-99/ftp/voip_protocols/">the original</a> on January 19, 2024<span class="reference-accessdate">. Retrieved <span class="nowrap">March 19,</span> 2024</span>. <q>Voice over IP (VOIP) uses the Internet Protocol (IP) to transmit voice as packets over an IP network. ...in the Internet, anyone can capture the packets meant for someone else. Some security can be provided by using encryption and tunneling. 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Retrieved <span class="nowrap">April 29,</span> 2012</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=MTALK-Readme&rft.pub=Sunsite.edu&rft_id=http%3A%2F%2Fsunsite.unc.edu%2Fpub%2FLinux%2Fapps%2Fsound%2Ftalk%2Fmtalk.README&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-99"><span class="mw-cite-backlink"><b><a href="#cite_ref-99">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFKeating" class="citation web cs1">Keating, Tom. <a rel="nofollow" class="external text" href="http://blog.tmcnet.com/blog/tom-keating/docs/cti-buyers-guide-1996.pdf">"Internet Phone Release 4"</a> <span class="cs1-format">(PDF)</span>. Computer Telephony Interaction Magazine<span class="reference-accessdate">. Retrieved <span class="nowrap">November 7,</span> 2007</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=Internet+Phone+Release+4&rft.pub=Computer+Telephony+Interaction+Magazine&rft.aulast=Keating&rft.aufirst=Tom&rft_id=http%3A%2F%2Fblog.tmcnet.com%2Fblog%2Ftom-keating%2Fdocs%2Fcti-buyers-guide-1996.pdf&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-100"><span class="mw-cite-backlink"><b><a href="#cite_ref-100">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite class="citation web cs1"><a rel="nofollow" class="external text" href="http://www.ilocus.com/2007/07/the_10_that_established_voip_p_2.html">"The 10 that Established VOIP (Part 1: VocalTec)"</a>. iLocus<span class="reference-accessdate">. Retrieved <span class="nowrap">January 21,</span> 2009</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=The+10+that+Established+VOIP+%28Part+1%3A+VocalTec%29&rft.pub=iLocus&rft_id=http%3A%2F%2Fwww.ilocus.com%2F2007%2F07%2Fthe_10_that_established_voip_p_2.html&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-101"><span class="mw-cite-backlink"><b><a href="#cite_ref-101">^</a></b></span> <span class="reference-text">The free Library <a rel="nofollow" class="external text" href="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.</a> <a rel="nofollow" class="external text" href="https://web.archive.org/web/20131030095918/http://www.thefreelibrary.com/RADVision+and+Intel+Target+Compatibility+Between+RADVision%27s+...-a019467970">Archived</a> October 30, 2013, at the <a href="/wiki/Wayback_Machine" title="Wayback Machine">Wayback Machine</a> June 2, 1997 <a rel="nofollow" class="external text" href="http://www.radvision.com/Developer-Solutions/">VoiP Developer Solutions</a> <a rel="nofollow" class="external text" href="https://web.archive.org/web/20110616001931/http://www.radvision.com/Developer-Solutions">Archived</a> June 16, 2011, at the <a href="/wiki/Wayback_Machine" title="Wayback Machine">Wayback Machine</a></span> </li> <li id="cite_note-102"><span class="mw-cite-backlink"><b><a href="#cite_ref-102">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite class="citation web cs1"><a rel="nofollow" class="external text" href="http://www.itu.int/rec/T-REC-H.323-199611-S/en">"H.323 Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service"</a>. ITU-T<span class="reference-accessdate">. Retrieved <span class="nowrap">January 21,</span> 2009</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=H.323+Visual+telephone+systems+and+equipment+for+local+area+networks+which+provide+a+non-guaranteed+quality+of+service&rft.pub=ITU-T&rft_id=http%3A%2F%2Fwww.itu.int%2Frec%2FT-REC-H.323-199611-S%2Fen&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-103"><span class="mw-cite-backlink"><b><a href="#cite_ref-103">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite class="citation web cs1"><a rel="nofollow" class="external text" href="http://www.faqs.org/rfcs/rfc2235.html">"RFC 2235"</a>. R. Zakon<span class="reference-accessdate">. Retrieved <span class="nowrap">January 21,</span> 2009</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=RFC+2235&rft.pub=R.+Zakon&rft_id=http%3A%2F%2Fwww.faqs.org%2Frfcs%2Frfc2235.html&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-104"><span class="mw-cite-backlink"><b><a href="#cite_ref-104">^</a></b></span> <span class="reference-text">International Telecommunication Union, Standardization Sector (ITU-T), Study Group 15 (1993-1996), <i>Recommendation G.729</i>, March 1996.</span> </li> <li id="cite_note-ilocus_softswitch-105"><span class="mw-cite-backlink"><b><a href="#cite_ref-ilocus_softswitch_105-0">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite class="citation web cs1"><a rel="nofollow" class="external text" href="http://www.ilocus.com/2007/07/the_10_that_established_voip_p_1.html">"The 10 that Established VOIP (Part 2: Level 3)"</a>. iLocus. July 13, 2007<span class="reference-accessdate">. Retrieved <span class="nowrap">November 7,</span> 2007</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=The+10+that+Established+VOIP+%28Part+2%3A+Level+3%29&rft.pub=iLocus&rft.date=2007-07-13&rft_id=http%3A%2F%2Fwww.ilocus.com%2F2007%2F07%2Fthe_10_that_established_voip_p_1.html&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-106"><span class="mw-cite-backlink"><b><a href="#cite_ref-106">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite class="citation web cs1"><a rel="nofollow" class="external text" href="http://www.ietf.org/rfc/rfc2543.txt">"RFC 2543, SIP: Session Initiation Protocol"</a>. Handley, Schulzrinne, Schooler, Rosenberg<span class="reference-accessdate">. Retrieved <span class="nowrap">January 21,</span> 2009</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=RFC+2543%2C+SIP%3A+Session+Initiation+Protocol&rft.pub=Handley%2C+Schulzrinne%2C+Schooler%2C+Rosenberg&rft_id=http%3A%2F%2Fwww.ietf.org%2Frfc%2Frfc2543.txt&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-107"><span class="mw-cite-backlink"><b><a href="#cite_ref-107">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite class="citation web cs1"><a rel="nofollow" class="external text" href="https://web.archive.org/web/20090123154417/http://www.asterisk.org/about">"What is Asterisk"</a>. Asterisk.org. Archived from <a rel="nofollow" class="external text" href="http://www.asterisk.org/about">the original</a> on January 23, 2009<span class="reference-accessdate">. Retrieved <span class="nowrap">January 21,</span> 2009</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=What+is+Asterisk&rft.pub=Asterisk.org&rft_id=http%3A%2F%2Fwww.asterisk.org%2Fabout&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-Hersent-108"><span class="mw-cite-backlink"><b><a href="#cite_ref-Hersent_108-0">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFHersentPetitGurle2005" class="citation book cs1">Hersent, Olivier; Petit, Jean-Pierre; Gurle, David (2005). <a rel="nofollow" class="external text" href="https://books.google.com/books?id=SMvNToRs-DgC&pg=PA55"><i>Beyond VoIP Protocols: Understanding Voice Technology and Networking Techniques for IP Telephony</i></a>. <a href="/wiki/John_Wiley_%26_Sons" class="mw-redirect" title="John Wiley & Sons">John Wiley & Sons</a>. p. 55. <a href="/wiki/ISBN_(identifier)" class="mw-redirect" title="ISBN (identifier)">ISBN</a> <a href="/wiki/Special:BookSources/9780470023631" title="Special:BookSources/9780470023631"><bdi>9780470023631</bdi></a>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=book&rft.btitle=Beyond+VoIP+Protocols%3A+Understanding+Voice+Technology+and+Networking+Techniques+for+IP+Telephony&rft.pages=55&rft.pub=John+Wiley+%26+Sons&rft.date=2005&rft.isbn=9780470023631&rft.aulast=Hersent&rft.aufirst=Olivier&rft.au=Petit%2C+Jean-Pierre&rft.au=Gurle%2C+David&rft_id=https%3A%2F%2Fbooks.google.com%2Fbooks%3Fid%3DSMvNToRs-DgC%26pg%3DPA55&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-Lutzky-109"><span class="mw-cite-backlink"><b><a href="#cite_ref-Lutzky_109-0">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFLutzkySchullerGayerKrämer2004" class="citation conference cs1">Lutzky, Manfred; Schuller, Gerald; Gayer, Marc; Krämer, Ulrich; Wabnik, Stefan (May 2004). <a rel="nofollow" class="external text" href="https://www.iis.fraunhofer.de/content/dam/iis/de/doc/ame/conference/AES-116-Convention_guideline-to-audio-codec-delay_AES116.pdf"><i>A guideline to audio codec delay</i></a> <span class="cs1-format">(PDF)</span>. 116th AES Convention. <i><a href="/wiki/Fraunhofer_IIS" class="mw-redirect" title="Fraunhofer IIS">Fraunhofer IIS</a></i>. <a href="/wiki/Audio_Engineering_Society" title="Audio Engineering Society">Audio Engineering Society</a><span class="reference-accessdate">. Retrieved <span class="nowrap">October 24,</span> 2019</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Ajournal&rft.genre=conference&rft.jtitle=Fraunhofer+IIS&rft.atitle=A+guideline+to+audio+codec+delay&rft.date=2004-05&rft.aulast=Lutzky&rft.aufirst=Manfred&rft.au=Schuller%2C+Gerald&rft.au=Gayer%2C+Marc&rft.au=Kr%C3%A4mer%2C+Ulrich&rft.au=Wabnik%2C+Stefan&rft_id=https%3A%2F%2Fwww.iis.fraunhofer.de%2Fcontent%2Fdam%2Fiis%2Fde%2Fdoc%2Fame%2Fconference%2FAES-116-Convention_guideline-to-audio-codec-delay_AES116.pdf&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-Schnell-110"><span class="mw-cite-backlink"><b><a href="#cite_ref-Schnell_110-0">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFSchnellSchmidtJanderAlbert2008" class="citation conference cs1">Schnell, Markus; Schmidt, Markus; Jander, Manuel; Albert, Tobias; Geiger, Ralf; Ruoppila, Vesa; Ekstrand, Per; Bernhard, Grill (October 2008). <a rel="nofollow" class="external text" href="https://www.iis.fraunhofer.de/content/dam/iis/de/doc/ame/conference/AES-125-Convention_AAC-ELD-NewStandardForHighQualityCommunication_AES7503.pdf"><i>MPEG-4 Enhanced Low Delay AAC - A New Standard for High Quality Communication</i></a> <span class="cs1-format">(PDF)</span>. 125th AES Convention. <i><a href="/wiki/Fraunhofer_IIS" class="mw-redirect" title="Fraunhofer IIS">Fraunhofer IIS</a></i>. <a href="/wiki/Audio_Engineering_Society" title="Audio Engineering Society">Audio Engineering Society</a><span class="reference-accessdate">. Retrieved <span class="nowrap">October 20,</span> 2019</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Ajournal&rft.genre=conference&rft.jtitle=Fraunhofer+IIS&rft.atitle=MPEG-4+Enhanced+Low+Delay+AAC+-+A+New+Standard+for+High+Quality+Communication&rft.date=2008-10&rft.aulast=Schnell&rft.aufirst=Markus&rft.au=Schmidt%2C+Markus&rft.au=Jander%2C+Manuel&rft.au=Albert%2C+Tobias&rft.au=Geiger%2C+Ralf&rft.au=Ruoppila%2C+Vesa&rft.au=Ekstrand%2C+Per&rft.au=Bernhard%2C+Grill&rft_id=https%3A%2F%2Fwww.iis.fraunhofer.de%2Fcontent%2Fdam%2Fiis%2Fde%2Fdoc%2Fame%2Fconference%2FAES-125-Convention_AAC-ELD-NewStandardForHighQualityCommunication_AES7503.pdf&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-111"><span class="mw-cite-backlink"><b><a href="#cite_ref-111">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFStapleton-Gray2009" class="citation book cs1">Stapleton-Gray, Ross (2009). <i>Inter-Network Operations Center Dial-by-ASN (INOC-DBA), a Resource for the Network Operator Community</i>. Los Alamitos: IEEE Computer Society Press. <a href="/wiki/ISBN_(identifier)" class="mw-redirect" title="ISBN (identifier)">ISBN</a> <a href="/wiki/Special:BookSources/978-0-7695-3568-5" title="Special:BookSources/978-0-7695-3568-5"><bdi>978-0-7695-3568-5</bdi></a>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=book&rft.btitle=Inter-Network+Operations+Center+Dial-by-ASN+%28INOC-DBA%29%2C+a+Resource+for+the+Network+Operator+Community&rft.place=Los+Alamitos&rft.pub=IEEE+Computer+Society+Press&rft.date=2009&rft.isbn=978-0-7695-3568-5&rft.aulast=Stapleton-Gray&rft.aufirst=Ross&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-112"><span class="mw-cite-backlink"><b><a href="#cite_ref-112">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFPogue2007" class="citation news cs1">Pogue, David (August 2, 2007). <a rel="nofollow" class="external text" href="https://www.nytimes.com/2007/08/02/technology/circuits/02pogue.html">"State of the Art: Get Your Free Net Phone Calls Here"</a>. <i>The New York Times</i><span class="reference-accessdate">. Retrieved <span class="nowrap">January 20,</span> 2009</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Ajournal&rft.genre=article&rft.jtitle=The+New+York+Times&rft.atitle=State+of+the+Art%3A+Get+Your+Free+Net+Phone+Calls+Here&rft.date=2007-08-02&rft.aulast=Pogue&rft.aufirst=David&rft_id=https%3A%2F%2Fwww.nytimes.com%2F2007%2F08%2F02%2Ftechnology%2Fcircuits%2F02pogue.html&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-Nagireddi-113"><span class="mw-cite-backlink"><b><a href="#cite_ref-Nagireddi_113-0">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFNagireddi2008" class="citation book cs1">Nagireddi, Sivannarayana (2008). <a rel="nofollow" class="external text" href="https://books.google.com/books?id=5AneeZFE71MC&pg=PA69"><i>VoIP Voice and Fax Signal Processing</i></a>. <a href="/wiki/John_Wiley_%26_Sons" class="mw-redirect" title="John Wiley & Sons">John Wiley & Sons</a>. p. 69. <a href="/wiki/ISBN_(identifier)" class="mw-redirect" title="ISBN (identifier)">ISBN</a> <a href="/wiki/Special:BookSources/9780470377864" title="Special:BookSources/9780470377864"><bdi>9780470377864</bdi></a>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=book&rft.btitle=VoIP+Voice+and+Fax+Signal+Processing&rft.pages=69&rft.pub=John+Wiley+%26+Sons&rft.date=2008&rft.isbn=9780470377864&rft.aulast=Nagireddi&rft.aufirst=Sivannarayana&rft_id=https%3A%2F%2Fbooks.google.com%2Fbooks%3Fid%3D5AneeZFE71MC%26pg%3DPA69&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-114"><span class="mw-cite-backlink"><b><a href="#cite_ref-114">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFRemo2007" class="citation news cs1">Remo, Michelle V. (August 27, 2007). <a rel="nofollow" class="external text" href="https://news.google.com/newspapers?nid=2479&dat=20070827&id=j1M1AAAAIBAJ&pg=1974,4860651">"Prospects bright for voice calls over internet"</a>. <i>Philippine Daily Inquirer</i><span class="reference-accessdate">. Retrieved <span class="nowrap">January 1,</span> 2015</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Ajournal&rft.genre=article&rft.jtitle=Philippine+Daily+Inquirer&rft.atitle=Prospects+bright+for+voice+calls+over+internet&rft.date=2007-08-27&rft.aulast=Remo&rft.aufirst=Michelle+V.&rft_id=https%3A%2F%2Fnews.google.com%2Fnewspapers%3Fnid%3D2479%26dat%3D20070827%26id%3Dj1M1AAAAIBAJ%26pg%3D1974%2C4860651&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-IETF79-115"><span class="mw-cite-backlink"><b><a href="#cite_ref-IETF79_115-0">^</a></b></span> <span class="reference-text"><a rel="nofollow" class="external text" href="http://nagasaki.bogus.com/ietf79/ietf79-ch8-tue-noon.mp3">Audio-Mitschnitt</a> <a rel="nofollow" class="external text" href="https://web.archive.org/web/20130210051956/http://nagasaki.bogus.com/ietf79/ietf79-ch8-tue-noon.mp3">Archived</a> February 10, 2013, at the <a href="/wiki/Wayback_Machine" title="Wayback Machine">Wayback Machine</a> vom Treffen der IETF-Codec-Arbeitsgruppe auf der Konferenz IETF79 in Peking, China mit einer Darstellung der grundlegenden Funktionsprinzipien durch Koen Vos (MP3, ~70 MiB)</span> </li> <li id="cite_note-116"><span class="mw-cite-backlink"><b><a href="#cite_ref-116">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite class="citation web cs1"><a rel="nofollow" class="external text" href="http://www.wirevolution.com/2009/01/13/skypes-new-super-wideband-codec/">"Skype's new super wideband codec"</a>. Wirevolution.com. January 13, 2009<span class="reference-accessdate">. Retrieved <span class="nowrap">March 31,</span> 2009</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=Skype%27s+new+super+wideband+codec&rft.pub=Wirevolution.com&rft.date=2009-01-13&rft_id=http%3A%2F%2Fwww.wirevolution.com%2F2009%2F01%2F13%2Fskypes-new-super-wideband-codec%2F&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-AppleInsider_standards_1-117"><span class="mw-cite-backlink"><b><a href="#cite_ref-AppleInsider_standards_1_117-0">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFDaniel_Eran_Dilger2010" class="citation web cs1">Daniel Eran Dilger (June 8, 2010). <a rel="nofollow" class="external text" href="http://www.appleinsider.com/articles/10/06/08/inside_iphone_4_facetime_video_calling.html">"Inside iPhone 4: FaceTime video calling"</a>. <a href="/wiki/Apple_community#AppleInsider" title="Apple community">AppleInsider</a><span class="reference-accessdate">. Retrieved <span class="nowrap">June 9,</span> 2010</span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=unknown&rft.btitle=Inside+iPhone+4%3A+FaceTime+video+calling&rft.pub=AppleInsider&rft.date=2010-06-08&rft.au=Daniel+Eran+Dilger&rft_id=http%3A%2F%2Fwww.appleinsider.com%2Farticles%2F10%2F06%2F08%2Finside_iphone_4_facetime_video_calling.html&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> <li id="cite_note-presentation-118"><span class="mw-cite-backlink"><b><a href="#cite_ref-presentation_118-0">^</a></b></span> <span class="reference-text"><a rel="nofollow" class="external text" href="http://people.xiph.org/~greg/video/linux_conf_au_CELT_2.ogv">Presentation of the CELT codec</a> <a rel="nofollow" class="external text" href="https://web.archive.org/web/20110807182250/http://people.xiph.org/~greg/video/linux_conf_au_CELT_2.ogv">Archived</a> August 7, 2011, at the <a href="/wiki/Wayback_Machine" title="Wayback Machine">Wayback Machine</a> by Timothy B. Terriberry (65 minutes of video, see also <a rel="nofollow" class="external text" href="http://www.celt-codec.org/presentations/misc/lca-celt.pdf">presentation slides</a> <a rel="nofollow" class="external text" href="https://web.archive.org/web/20110810032741/http://www.celt-codec.org/presentations/misc/lca-celt.pdf">Archived</a> August 10, 2011, at the <a href="/wiki/Wayback_Machine" title="Wayback Machine">Wayback Machine</a> in PDF)</span> </li> <li id="cite_note-Valin-119"><span class="mw-cite-backlink"><b><a href="#cite_ref-Valin_119-0">^</a></b></span> <span class="reference-text"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1238218222"><cite id="CITEREFValinMaxwellTerriberryVos2013" class="citation conference cs1">Valin, Jean-Marc; Maxwell, Gregory; Terriberry, Timothy B.; Vos, Koen (October 2013). <i>High-Quality, Low-Delay Music Coding in the Opus Codec</i>. 135th AES Convention. <a href="/wiki/Audio_Engineering_Society" title="Audio Engineering Society">Audio Engineering Society</a>. <a href="/wiki/ArXiv_(identifier)" class="mw-redirect" title="ArXiv (identifier)">arXiv</a>:<span class="id-lock-free" title="Freely accessible"><a rel="nofollow" class="external text" href="https://arxiv.org/abs/1602.04845">1602.04845</a></span>.</cite><span title="ctx_ver=Z39.88-2004&rft_val_fmt=info%3Aofi%2Ffmt%3Akev%3Amtx%3Abook&rft.genre=conference&rft.btitle=High-Quality%2C+Low-Delay+Music+Coding+in+the+Opus+Codec&rft.pub=Audio+Engineering+Society&rft.date=2013-10&rft_id=info%3Aarxiv%2F1602.04845&rft.aulast=Valin&rft.aufirst=Jean-Marc&rft.au=Maxwell%2C+Gregory&rft.au=Terriberry%2C+Timothy+B.&rft.au=Vos%2C+Koen&rfr_id=info%3Asid%2Fen.wikipedia.org%3AVoice+over+IP" class="Z3988"></span></span> </li> </ol></div></div> <div class="mw-heading mw-heading2"><h2 id="External_links">External links</h2><span class="mw-editsection"><span class="mw-editsection-bracket">[</span><a href="/w/index.php?title=Voice_over_IP&action=edit&section=39" title="Edit section: External links"><span>edit</span></a><span class="mw-editsection-bracket">]</span></span></div> <ul><li><span class="noviewer" typeof="mw:File"><a href="/wiki/File:Wiktionary-logo-en-v2.svg" class="mw-file-description"><img alt="" src="//upload.wikimedia.org/wikipedia/commons/thumb/9/99/Wiktionary-logo-en-v2.svg/16px-Wiktionary-logo-en-v2.svg.png" decoding="async" width="16" height="16" class="mw-file-element" srcset="//upload.wikimedia.org/wikipedia/commons/thumb/9/99/Wiktionary-logo-en-v2.svg/24px-Wiktionary-logo-en-v2.svg.png 1.5x, //upload.wikimedia.org/wikipedia/commons/thumb/9/99/Wiktionary-logo-en-v2.svg/32px-Wiktionary-logo-en-v2.svg.png 2x" data-file-width="512" data-file-height="512" /></a></span> The dictionary definition of <a href="https://en.wiktionary.org/wiki/VoIP" class="extiw" title="wiktionary:VoIP"><i>VoIP</i></a> at Wiktionary</li> <li><span class="noviewer" typeof="mw:File"><a href="/wiki/File:Wikivoyage-Logo-v3-icon.svg" class="mw-file-description"><img alt="" src="//upload.wikimedia.org/wikipedia/commons/thumb/d/dd/Wikivoyage-Logo-v3-icon.svg/16px-Wikivoyage-Logo-v3-icon.svg.png" decoding="async" width="16" height="16" class="mw-file-element" srcset="//upload.wikimedia.org/wikipedia/commons/thumb/d/dd/Wikivoyage-Logo-v3-icon.svg/24px-Wikivoyage-Logo-v3-icon.svg.png 1.5x, //upload.wikimedia.org/wikipedia/commons/thumb/d/dd/Wikivoyage-Logo-v3-icon.svg/32px-Wikivoyage-Logo-v3-icon.svg.png 2x" data-file-width="193" data-file-height="193" /></a></span> <a href="https://en.wikivoyage.org/wiki/Internet_telephony#Q81945" class="extiw" title="voy:Internet telephony">Internet telephony</a> travel guide from Wikivoyage</li></ul> <div class="navbox-styles"><style data-mw-deduplicate="TemplateStyles:r1129693374">.mw-parser-output .hlist dl,.mw-parser-output .hlist ol,.mw-parser-output .hlist ul{margin:0;padding:0}.mw-parser-output .hlist dd,.mw-parser-output .hlist dt,.mw-parser-output .hlist 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mailing list</a></li> <li><a href="/wiki/FidoNet" title="FidoNet">FidoNet</a></li> <li><a href="/wiki/Usenet" title="Usenet">Usenet</a></li> <li><a href="/wiki/Internet_forum" title="Internet forum">Internet forum</a> <ul><li><a href="/wiki/Textboard" title="Textboard">Textboard</a></li> <li><a href="/wiki/Imageboard" title="Imageboard">Imageboard</a></li></ul></li> <li><a href="/wiki/Bulletin_board_system" title="Bulletin board system">Bulletin board system</a></li> <li><a href="/wiki/Guestbook" title="Guestbook">Online guestbook</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%"><a href="/wiki/Synchronous_conferencing" class="mw-redirect" title="Synchronous conferencing">Synchronous conferencing</a></th><td class="navbox-list-with-group navbox-list navbox-even" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Data_conferencing" title="Data conferencing">Data conferencing</a></li> <li><a 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href="/wiki/Fiber-optic_communication" title="Fiber-optic communication">Fibre-optical</a></li> <li><a href="/wiki/Free-space_optical_communication" title="Free-space optical communication">Free-space optical</a></li> <li><a href="/wiki/Integrated_Services_Digital_Network" class="mw-redirect" title="Integrated Services Digital Network">ISDN</a></li> <li><a href="/wiki/Mobile_phone_signal" title="Mobile phone signal">Mobile phone signal</a></li> <li><a href="/wiki/Plain_old_telephone_service" title="Plain old telephone service">POTS</a></li> <li><a href="/wiki/Public_switched_telephone_network" title="Public switched telephone network">PSTN</a></li> <li><a href="/wiki/Submarine_communications_cable" title="Submarine communications cable">Submarine cables</a></li> <li><a class="mw-selflink selflink">VoIP</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%">Calls</th><td class="navbox-list-with-group navbox-list navbox-even hlist" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Missed_call" title="Missed call">Missed call</a></li> <li><a href="/wiki/Misdialed_call" title="Misdialed call">Misdialed call</a></li> <li><a href="/wiki/Nuisance_call" title="Nuisance call">Nuisance call</a></li> <li><a href="/wiki/Phone_tag" title="Phone tag">Phone tag</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%">Applications</th><td class="navbox-list-with-group navbox-list navbox-odd hlist" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Fax" title="Fax">Fax transmission</a></li> <li><a href="/wiki/Telephone_call" title="Telephone call">Telephone calls</a></li> <li><a href="/wiki/Telephone_newspaper" title="Telephone newspaper">Telephone newspapers</a></li> <li><a href="/wiki/Th%C3%A9%C3%A2trophone" title="Théâtrophone">Théâtrophone</a></li> <li><a href="/wiki/Videotelephony" title="Videotelephony">Video calls</a></li></ul> </div></td></tr><tr><td class="navbox-abovebelow" colspan="2"><div><span class="nowrap"><span class="noviewer" typeof="mw:File"><span><img alt="" src="//upload.wikimedia.org/wikipedia/commons/thumb/4/4e/Telecom-icon.svg/28px-Telecom-icon.svg.png" decoding="async" width="28" height="28" class="mw-file-element" srcset="//upload.wikimedia.org/wikipedia/commons/thumb/4/4e/Telecom-icon.svg/42px-Telecom-icon.svg.png 1.5x, //upload.wikimedia.org/wikipedia/commons/thumb/4/4e/Telecom-icon.svg/56px-Telecom-icon.svg.png 2x" data-file-width="500" data-file-height="500" /></span></span> </span><a href="/wiki/Portal:Telecommunication" title="Portal:Telecommunication">Telecommunication portal</a> <span class="nowrap"><span class="noviewer" typeof="mw:File"><span><img alt="telephone icon" src="//upload.wikimedia.org/wikipedia/commons/thumb/b/b8/Telephone_icon_blue_gradient.svg/28px-Telephone_icon_blue_gradient.svg.png" decoding="async" width="28" height="28" class="mw-file-element" srcset="//upload.wikimedia.org/wikipedia/commons/thumb/b/b8/Telephone_icon_blue_gradient.svg/42px-Telephone_icon_blue_gradient.svg.png 1.5x, //upload.wikimedia.org/wikipedia/commons/thumb/b/b8/Telephone_icon_blue_gradient.svg/56px-Telephone_icon_blue_gradient.svg.png 2x" data-file-width="722" data-file-height="722" /></span></span> </span><a href="/wiki/Portal:Telephones" title="Portal:Telephones">Telephones portal</a></div></td></tr></tbody></table></div> <div class="navbox-styles"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1129693374"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1236075235"></div><div role="navigation" class="navbox" aria-labelledby="Telecommunications" style="padding:3px"><table class="nowraplinks hlist mw-collapsible mw-collapsed navbox-inner" style="border-spacing:0;background:transparent;color:inherit"><tbody><tr><th scope="col" class="navbox-title" colspan="2"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1129693374"><link rel="mw-deduplicated-inline-style" href="mw-data:TemplateStyles:r1239400231"><div class="navbar plainlinks hlist navbar-mini"><ul><li class="nv-view"><a href="/wiki/Template:Telecommunications" title="Template:Telecommunications"><abbr title="View this template">v</abbr></a></li><li class="nv-talk"><a href="/wiki/Template_talk:Telecommunications" title="Template talk:Telecommunications"><abbr title="Discuss this template">t</abbr></a></li><li class="nv-edit"><a href="/wiki/Special:EditPage/Template:Telecommunications" title="Special:EditPage/Template:Telecommunications"><abbr title="Edit this template">e</abbr></a></li></ul></div><div id="Telecommunications" style="font-size:114%;margin:0 4em"><a href="/wiki/Telecommunications" title="Telecommunications">Telecommunications</a></div></th></tr><tr><th scope="row" class="navbox-group" style="width:1%"><a href="/wiki/History_of_telecommunication" title="History of telecommunication">History</a></th><td class="navbox-list-with-group navbox-list navbox-odd" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Beacon#For_defensive_communications" title="Beacon">Beacon</a></li> <li><a href="/wiki/History_of_broadcasting" title="History of broadcasting">Broadcasting</a></li> <li><a href="/wiki/Cable_protection_system" title="Cable protection system">Cable protection system</a></li> <li><a href="/wiki/Cable_television" title="Cable television">Cable TV</a></li> <li><a href="/wiki/Communications_satellite#History" title="Communications satellite">Communications satellite</a></li> <li><a href="/wiki/Computer_network#History" title="Computer network">Computer network</a></li> <li><a href="/wiki/Data_compression" title="Data compression">Data compression</a> <ul><li><a href="/wiki/Audio_coding_format" title="Audio coding format">audio</a></li> <li><a href="/wiki/Discrete_cosine_transform" title="Discrete cosine transform">DCT</a></li> <li><a href="/wiki/Image_compression" title="Image compression">image</a></li> <li><a href="/wiki/Video_coding_format" title="Video coding format">video</a></li></ul></li> <li><a href="/wiki/Digital_media" title="Digital media">Digital media</a> <ul><li><a href="/wiki/Internet_video" title="Internet video">Internet video</a></li> <li><a href="/wiki/Online_video_platform" title="Online video platform">online video platform</a></li> <li><a href="/wiki/Social_media" title="Social media">social media</a></li> <li><a href="/wiki/Streaming_media" title="Streaming media">streaming</a></li></ul></li> <li><a href="/wiki/Drums_in_communication" title="Drums in communication">Drums</a></li> <li><a href="/wiki/Edholm%27s_law" title="Edholm's law">Edholm's law</a></li> <li><a href="/wiki/Electrical_telegraph#History" title="Electrical telegraph">Electrical telegraph</a></li> <li><a href="/wiki/Fax#History" title="Fax">Fax</a></li> <li><a href="/wiki/Heliograph#History" title="Heliograph">Heliographs</a></li> <li><a href="/wiki/Hydraulic_telegraph#Greek_hydraulic_semaphore_system" title="Hydraulic telegraph">Hydraulic telegraph</a></li> <li><a href="/wiki/Information_Age" title="Information Age">Information Age</a></li> <li><a href="/wiki/Information_revolution" class="mw-redirect" title="Information revolution">Information revolution</a></li> <li><a href="/wiki/History_of_the_Internet" title="History of the Internet">Internet</a></li> <li><a href="/wiki/Mass_media#History" title="Mass media">Mass media</a></li> <li><a href="/wiki/History_of_mobile_phones" title="History of mobile phones">Mobile phone</a> <ul><li><a href="/wiki/Smartphone" title="Smartphone">Smartphone</a></li></ul></li> <li><a href="/wiki/Optical_communication" title="Optical communication">Optical telecommunication</a></li> <li><a href="/wiki/Optical_telegraph" title="Optical telegraph">Optical telegraphy</a></li> <li><a href="/wiki/Pager" title="Pager">Pager</a></li> <li><a href="/wiki/Photophone" title="Photophone">Photophone</a></li> <li><a href="/wiki/History_of_prepaid_mobile_phones" title="History of prepaid mobile phones">Prepaid mobile phone</a></li> <li><a href="/wiki/History_of_radio" title="History of radio">Radio</a></li> <li><a href="/wiki/Radiotelephone" title="Radiotelephone">Radiotelephone</a></li> <li><a href="/wiki/Communications_satellite" title="Communications satellite">Satellite communications</a></li> <li><a href="/wiki/Semaphore" title="Semaphore">Semaphore</a> <ul><li><a href="/wiki/Phryctoria" title="Phryctoria">Phryctoria</a></li></ul></li> <li><a href="/wiki/Semiconductor" title="Semiconductor">Semiconductor</a> <ul><li><a href="/wiki/Semiconductor_device" title="Semiconductor device">device</a></li> <li><a href="/wiki/MOSFET" title="MOSFET">MOSFET</a></li> <li><a href="/wiki/History_of_the_transistor" title="History of the transistor">transistor</a></li></ul></li> <li><a href="/wiki/Smoke_signal" title="Smoke signal">Smoke signals</a></li> <li><a href="/wiki/History_of_telecommunication" title="History of telecommunication">Telecommunications history</a></li> <li><a href="/wiki/Telautograph" title="Telautograph">Telautograph</a></li> <li><a href="/wiki/Telegraphy" title="Telegraphy">Telegraphy</a></li> <li><a href="/wiki/Teleprinter" title="Teleprinter">Teleprinter</a> (teletype)</li> <li><a href="/wiki/History_of_the_telephone" title="History of the telephone">Telephone</a></li> <li><i><a href="/wiki/The_Telephone_Cases" title="The Telephone Cases">The Telephone Cases</a></i></li> <li><a href="/wiki/History_of_television" title="History of television">Television</a> <ul><li><a href="/wiki/Digital_television" title="Digital television">digital</a></li> <li><a href="/wiki/Streaming_television" title="Streaming television">streaming</a></li></ul></li> <li><a href="/wiki/Submarine_communications_cable#Early_history:_telegraph_and_coaxial_cables" title="Submarine communications cable">Undersea telegraph line</a></li> <li><a href="/wiki/History_of_videotelephony" title="History of videotelephony">Videotelephony</a></li> <li><a href="/wiki/Whistled_language" title="Whistled language">Whistled language</a></li> <li><a href="/wiki/Wireless_revolution" class="mw-redirect" title="Wireless revolution">Wireless revolution</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%">Pioneers</th><td class="navbox-list-with-group navbox-list navbox-even" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Nasir_Ahmed_(engineer)" title="Nasir Ahmed (engineer)">Nasir Ahmed</a></li> <li><a href="/wiki/Edwin_Howard_Armstrong" title="Edwin Howard Armstrong">Edwin Howard Armstrong</a></li> <li><a href="/wiki/Mohamed_M._Atalla" title="Mohamed M. Atalla">Mohamed M. Atalla</a></li> <li><a href="/wiki/John_Logie_Baird" title="John Logie Baird">John Logie Baird</a></li> <li><a href="/wiki/Paul_Baran" title="Paul Baran">Paul Baran</a></li> <li><a href="/wiki/John_Bardeen" title="John Bardeen">John Bardeen</a></li> <li><a href="/wiki/Alexander_Graham_Bell" title="Alexander Graham Bell">Alexander Graham Bell</a></li> <li><a href="/wiki/Emile_Berliner" title="Emile Berliner">Emile Berliner</a></li> <li><a href="/wiki/Tim_Berners-Lee" title="Tim Berners-Lee">Tim Berners-Lee</a></li> <li><a href="/wiki/Francis_Blake_(inventor)" title="Francis Blake (inventor)">Francis Blake</a></li> <li><a href="/wiki/Jagadish_Chandra_Bose" title="Jagadish Chandra Bose">Jagadish Chandra Bose</a></li> <li><a href="/wiki/Charles_Bourseul" title="Charles Bourseul">Charles Bourseul</a></li> <li><a href="/wiki/Walter_Houser_Brattain" title="Walter Houser Brattain">Walter Houser Brattain</a></li> <li><a href="/wiki/Vint_Cerf" title="Vint Cerf">Vint Cerf</a></li> <li><a href="/wiki/Claude_Chappe" title="Claude Chappe">Claude Chappe</a></li> <li><a href="/wiki/Yogen_Dalal" class="mw-redirect" title="Yogen Dalal">Yogen Dalal</a></li> <li><a href="/wiki/Daniel_Davis_Jr." title="Daniel Davis Jr.">Daniel Davis Jr.</a></li> <li><a href="/wiki/Donald_Davies" title="Donald Davies">Donald Davies</a></li> <li><a href="/wiki/Amos_Dolbear" title="Amos Dolbear">Amos Dolbear</a></li> <li><a href="/wiki/Thomas_Edison" title="Thomas Edison">Thomas Edison</a></li> <li><a href="/wiki/Lee_de_Forest" title="Lee de Forest">Lee de Forest</a></li> <li><a href="/wiki/Philo_Farnsworth" title="Philo Farnsworth">Philo Farnsworth</a></li> <li><a href="/wiki/Reginald_Fessenden" title="Reginald Fessenden">Reginald Fessenden</a></li> <li><a href="/wiki/Elisha_Gray" title="Elisha Gray">Elisha Gray</a></li> <li><a href="/wiki/Oliver_Heaviside" title="Oliver Heaviside">Oliver Heaviside</a></li> <li><a href="/wiki/Robert_Hooke" title="Robert Hooke">Robert Hooke</a></li> <li><a href="/wiki/Erna_Schneider_Hoover" title="Erna Schneider Hoover">Erna Schneider Hoover</a></li> <li><a href="/wiki/Harold_Hopkins_(physicist)" title="Harold Hopkins (physicist)">Harold Hopkins</a></li> <li><a href="/wiki/Gardiner_Greene_Hubbard" title="Gardiner Greene Hubbard">Gardiner Greene Hubbard</a></li> <li><a href="/wiki/List_of_Internet_pioneers" title="List of Internet pioneers">Internet pioneers</a></li> <li><a href="/wiki/Bob_Kahn" class="mw-redirect" title="Bob Kahn">Bob Kahn</a></li> <li><a href="/wiki/Dawon_Kahng" title="Dawon Kahng">Dawon Kahng</a></li> <li><a href="/wiki/Charles_K._Kao" title="Charles K. Kao">Charles K. Kao</a></li> <li><a href="/wiki/Narinder_Singh_Kapany" title="Narinder Singh Kapany">Narinder Singh Kapany</a></li> <li><a href="/wiki/Hedy_Lamarr" title="Hedy Lamarr">Hedy Lamarr</a></li> <li><a href="/wiki/Roberto_Landell_de_Moura" title="Roberto Landell de Moura">Roberto Landell de Moura</a></li> <li><a href="/wiki/Innocenzo_Manzetti" title="Innocenzo Manzetti">Innocenzo Manzetti</a></li> <li><a href="/wiki/Guglielmo_Marconi" title="Guglielmo Marconi">Guglielmo Marconi</a></li> <li><a href="/wiki/Robert_Metcalfe" title="Robert Metcalfe">Robert Metcalfe</a></li> <li><a href="/wiki/Antonio_Meucci" title="Antonio Meucci">Antonio Meucci</a></li> <li><a href="/wiki/Samuel_Morse" title="Samuel Morse">Samuel Morse</a></li> <li><a href="/wiki/Jun-ichi_Nishizawa" title="Jun-ichi Nishizawa">Jun-ichi Nishizawa</a></li> <li><a href="/wiki/Charles_Grafton_Page" title="Charles Grafton Page">Charles Grafton Page</a></li> <li><a href="/wiki/Radia_Perlman" title="Radia Perlman">Radia Perlman</a></li> <li><a href="/wiki/Alexander_Stepanovich_Popov" class="mw-redirect" title="Alexander Stepanovich Popov">Alexander Stepanovich Popov</a></li> <li><a href="/wiki/Tivadar_Pusk%C3%A1s" title="Tivadar Puskás">Tivadar Puskás</a></li> <li><a href="/wiki/Johann_Philipp_Reis" title="Johann Philipp Reis">Johann Philipp Reis</a></li> <li><a href="/wiki/Claude_Shannon" title="Claude Shannon">Claude Shannon</a></li> <li><a href="/wiki/Almon_Brown_Strowger" title="Almon Brown Strowger">Almon Brown Strowger</a></li> <li><a href="/wiki/Henry_Sutton_(inventor)" title="Henry Sutton (inventor)">Henry Sutton</a></li> <li><a href="/wiki/Charles_Sumner_Tainter" title="Charles Sumner Tainter">Charles Sumner Tainter</a></li> <li><a href="/wiki/Nikola_Tesla" title="Nikola Tesla">Nikola Tesla</a></li> <li><a href="/wiki/Camille_Tissot" title="Camille Tissot">Camille Tissot</a></li> <li><a href="/wiki/Alfred_Vail" title="Alfred Vail">Alfred Vail</a></li> <li><a href="/wiki/Thomas_A._Watson" title="Thomas A. Watson">Thomas A. Watson</a></li> <li><a href="/wiki/Charles_Wheatstone" title="Charles Wheatstone">Charles Wheatstone</a></li> <li><a href="/wiki/Vladimir_K._Zworykin" title="Vladimir K. Zworykin">Vladimir K. Zworykin</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%"><a href="/wiki/Transmission_medium" title="Transmission medium">Transmission<br />media</a></th><td class="navbox-list-with-group navbox-list navbox-odd" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Coaxial_cable" title="Coaxial cable">Coaxial cable</a></li> <li><a href="/wiki/Fiber-optic_communication" title="Fiber-optic communication">Fiber-optic communication</a> <ul><li><a href="/wiki/Optical_fiber" title="Optical fiber">optical fiber</a></li></ul></li> <li><a href="/wiki/Free-space_optical_communication" title="Free-space optical communication">Free-space optical communication</a></li> <li><a href="/wiki/Molecular_communication" title="Molecular communication">Molecular communication</a></li> <li><a href="/wiki/Radio_wave" title="Radio wave">Radio waves</a> <ul><li><a href="/wiki/Wireless" title="Wireless">wireless</a></li></ul></li> <li><a href="/wiki/Transmission_line" title="Transmission line">Transmission line</a> <ul><li><a href="/wiki/Telecommunication_circuit" title="Telecommunication circuit">telecommunication circuit</a></li></ul></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%"><a href="/wiki/Network_topology" title="Network topology">Network topology</a><br />and switching</th><td class="navbox-list-with-group navbox-list navbox-even" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Bandwidth_(computing)" title="Bandwidth (computing)">Bandwidth</a></li> <li><a href="/wiki/Telecommunications_link" title="Telecommunications link">Links</a></li> <li><a href="/wiki/Node_(networking)" title="Node (networking)">Nodes</a> <ul><li><a href="/wiki/Terminal_(telecommunication)" title="Terminal (telecommunication)">terminal</a></li></ul></li> <li><a href="/wiki/Network_switch" title="Network switch">Network switching</a> <ul><li><a href="/wiki/Circuit_switching" title="Circuit switching">circuit</a></li> <li><a href="/wiki/Packet_switching" title="Packet switching">packet</a></li></ul></li> <li><a href="/wiki/Telephone_exchange" title="Telephone exchange">Telephone exchange</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%"><a href="/wiki/Multiplexing" title="Multiplexing">Multiplexing</a></th><td class="navbox-list-with-group navbox-list navbox-odd" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Space-division_multiple_access" title="Space-division multiple access">Space-division</a></li> <li><a href="/wiki/Frequency-division_multiplexing" title="Frequency-division multiplexing">Frequency-division</a></li> <li><a href="/wiki/Time-division_multiplexing" title="Time-division multiplexing">Time-division</a></li> <li><a href="/wiki/Polarization-division_multiplexing" title="Polarization-division multiplexing">Polarization-division</a></li> <li><a href="/wiki/Orbital_angular_momentum_multiplexing" title="Orbital angular momentum multiplexing">Orbital angular-momentum</a></li> <li><a href="/wiki/Code-division_multiple_access" title="Code-division multiple access">Code-division</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%">Concepts</th><td class="navbox-list-with-group navbox-list navbox-even" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Communication_protocol" title="Communication protocol">Communication protocol</a></li> <li><a href="/wiki/Computer_network" title="Computer network">Computer network</a></li> <li><a href="/wiki/Data_communication" title="Data communication">Data transmission</a></li> <li><a href="/wiki/Store_and_forward" title="Store and forward">Store and forward</a></li> <li><a href="/wiki/Telecommunications_equipment" title="Telecommunications equipment">Telecommunications equipment</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%"><a href="/wiki/Telecommunications_network" title="Telecommunications network">Types of network</a></th><td class="navbox-list-with-group navbox-list navbox-odd" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Cellular_network" title="Cellular network">Cellular network</a></li> <li><a href="/wiki/Ethernet" title="Ethernet">Ethernet</a></li> <li><a href="/wiki/Integrated_Services_Digital_Network" class="mw-redirect" title="Integrated Services Digital Network">ISDN</a></li> <li><a href="/wiki/Local_area_network" title="Local area network">LAN</a></li> <li><a href="/wiki/Mobile_telephony" title="Mobile telephony">Mobile</a></li> <li><a href="/wiki/Next-generation_network" title="Next-generation network">NGN</a></li> <li><a href="/wiki/Public_switched_telephone_network" title="Public switched telephone network">Public Switched Telephone</a></li> <li><a href="/wiki/Radio_network" title="Radio network">Radio</a></li> <li><a href="/wiki/Television_broadcasting" class="mw-redirect" title="Television broadcasting">Television</a></li> <li><a href="/wiki/Telex" title="Telex">Telex</a></li> <li><a href="/wiki/UUCP" title="UUCP">UUCP</a></li> <li><a href="/wiki/Wide_area_network" title="Wide area network">WAN</a></li> <li><a href="/wiki/Wireless_network" title="Wireless network">Wireless network</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%"><a href="/wiki/Telecommunications_network" title="Telecommunications network">Notable networks</a></th><td class="navbox-list-with-group navbox-list navbox-even" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/ARPANET" title="ARPANET">ARPANET</a></li> <li><a href="/wiki/BITNET" title="BITNET">BITNET</a></li> <li><a href="/wiki/CYCLADES" title="CYCLADES">CYCLADES</a></li> <li><a href="/wiki/FidoNet" title="FidoNet">FidoNet</a></li> <li><a href="/wiki/Internet" title="Internet">Internet</a></li> <li><a href="/wiki/Internet2" title="Internet2">Internet2</a></li> <li><a href="/wiki/JANET" title="JANET">JANET</a></li> <li><a href="/wiki/NPL_network" title="NPL network">NPL network</a></li> <li><a href="/wiki/Toasternet" title="Toasternet">Toasternet</a></li> <li><a href="/wiki/Usenet" title="Usenet">Usenet</a></li></ul> </div></td></tr><tr><th scope="row" class="navbox-group" style="width:1%">Locations</th><td class="navbox-list-with-group navbox-list navbox-odd" style="width:100%;padding:0"><div style="padding:0 0.25em"> <ul><li><a href="/wiki/Category:Telecommunications_in_Africa" title="Category:Telecommunications in Africa">Africa</a></li> <li>Americas <ul><li><a href="/wiki/Category:Telecommunications_in_North_America" title="Category:Telecommunications in North America">North</a></li> <li><a href="/wiki/Category:Telecommunications_in_South_America" title="Category:Telecommunications in South America">South</a></li></ul></li> <li><a href="/wiki/Category:Communications_in_Antarctica" title="Category:Communications in Antarctica">Antarctica</a></li> <li><a href="/wiki/Category:Telecommunications_in_Asia" title="Category:Telecommunications in Asia">Asia</a></li> <li><a href="/wiki/Category:Telecommunications_in_Europe" title="Category:Telecommunications in Europe">Europe</a></li> <li><a href="/wiki/Category:Telecommunications_in_Oceania" title="Category:Telecommunications in Oceania">Oceania</a></li> <li>(<a href="/wiki/List_of_telecommunications_regulatory_bodies" title="List of telecommunications regulatory bodies">Global telecommunications regulation bodies</a>)</li></ul> </div></td></tr><tr><td class="navbox-abovebelow" colspan="2"><div> <ul><li><span class="nowrap"><span class="noviewer" typeof="mw:File"><span><img alt="" src="//upload.wikimedia.org/wikipedia/commons/thumb/4/4e/Telecom-icon.svg/16px-Telecom-icon.svg.png" decoding="async" width="16" height="16" class="mw-file-element" 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