{"id":26042,"date":"2021-04-20T13:22:38","date_gmt":"2021-04-20T12:22:38","guid":{"rendered":"https:\/\/voipstudio.com\/blog\/causas-soluciones-corte-de-llamadas-en-voip\/"},"modified":"2025-10-10T13:05:50","modified_gmt":"2025-10-10T12:05:50","slug":"causas-soluciones-corte-de-llamadas-en-voip","status":"publish","type":"post","link":"https:\/\/voipstudio.com\/es\/blog\/causas-soluciones-corte-de-llamadas-en-voip\/","title":{"rendered":"Causas y soluciones al corte de llamadas en VoIP"},"content":{"rendered":"<p><span style=\"font-weight: 400;\">En ocasiones al recibir una llamada, esta se establece correctamente, y <strong>mientras estamos hablando<\/strong> con el otro interlocutor a los pocos segundos <strong>se corta<\/strong>. Normalmente esta interrupci\u00f3n repentina suele producirse sobre los 30 segundos.\u00a0<\/span><\/p>\n<h2>Causas del corte de llamada VoIP<\/h2>\n<p><span style=\"font-weight: 400;\">Esto puede ocurrir cuando <strong>utilizamos softphones o terminales IP con<\/strong> protocolo de inicio de sesi\u00f3n <strong>SIP<\/strong>. El motivo de que ocurra esto es que seg\u00fan protocolo SIP el tel\u00e9fono debe responder con un ACK para indicar que ha recibido y contestado a la llamada entrante que recibe desde la centralita. <\/span><\/p>\n<p><span style=\"font-weight: 400;\">De esta manera la centralita sabe que la comunicaci\u00f3n est\u00e1 en curso. Si <strong>la centralita no recibe dicha confirmaci\u00f3n<\/strong> puede llegar a cortar la llamada al cabo de alg\u00fan tiempo.\u00a0<\/span><\/p>\n<p><span style=\"font-weight: 400;\">Existen diferentes motivos por los cuales esa respuesta de confirmaci\u00f3n que emite el terminal SIP hacia la centralita no alcanza la misma. El <strong>problema m\u00e1s com\u00fan<\/strong> es que algunos routers o firewall de operador implementan una <a href=\"https:\/\/www.watchguard.com\/help\/docs\/help-center\/es-419\/Content\/es-419\/Fireware\/proxies\/sip\/sip_proxy_about_c.html?TocPath=Controlar%20el%20Tr%C3%A1fico%20de%20Red%7CServidores%20Proxy%7CAcerca%20de%20SIP-ALG%7C_____0\" target=\"_blank\" rel=\"noopener\">funcionalidad que se llama \u201cSIP ALG<\/a>\u201d. Parad\u00f3jicamente esta funcionalidad \u201cSIP ALG\u201d dise\u00f1ada para mejorar la funci\u00f3n de NAT en la comunicaci\u00f3n SIP lo que hace en muchos casos es <strong>romper el protocolo SIP<\/strong>.<\/span><\/p>\n<h3><strong> Problemas causados al romperse el Protocolo SIP<\/strong><\/h3>\n<ul>\n<li><span style=\"font-weight: 400;\">Cortes de llamada a los pocos segundos.<\/span><\/li>\n<li><span style=\"font-weight: 400;\">Llamadas entrantes que no suenan, o no alcanzan el terminal.<\/span><\/li>\n<li><span style=\"font-weight: 400;\">Audio en un solo sentido.<\/span><\/li>\n<\/ul>\n<h3>Soluciones<\/h3>\n<p><span style=\"font-weight: 400;\">Para solventar esto lo mejor ser\u00eda<strong> desactivar el protocolo de SIP ALG<\/strong> en aquellos routers o firewalls donde sea posible. Para ello debemos poder administrar esos equipos, y estos tiene que tener accesible la activaci\u00f3n o no de esta funcionalidad. <\/span><\/p>\n<p><span style=\"font-weight: 400;\">Lamentablemente en la mayor\u00eda de casos no tenemos control sobre estos equipos o bien estos <strong>no tienen la funcionalidad disponible<\/strong>.\u00a0<\/span><\/p>\n<p><span style=\"font-weight: 400;\">Algunas otras soluciones que podemos adoptar en los terminales para intentar salvar este problema son:<\/span><\/p>\n<ol>\n<li><span style=\"font-weight: 400;\">Utilizar los <a href=\"https:\/\/voipstudio.com\/es\/descargas\/\">Softphones de VoIPstudio<\/a> para Windows, MAC, Linux, iPhone o Android donde sea posible. Nuestros softphones<strong> utilizan el protocolo WebRTC<\/strong> con el que evitar\u00edamos el problema descrito para SIP.<\/span><\/li>\n<li><span style=\"font-weight: 400;\">Habilitar y configurar el servidor de STUN de VoIPstudio \u201cstun.ssl7.net\u201d en aquellos equipos SIP que dispongan de esta opci\u00f3n. Por ejemplo, hemos visto que el softphone de Zoiper para smartphone es muy sensible a este problema. En la mayor\u00eda de casos configurando el STUN de VoIPstudio tal como se muestra en la imagen queda resuelto.<\/span><\/li>\n<\/ol>\n<p><img fetchpriority=\"high\" decoding=\"async\" class=\"aligncenter wp-image-40282 size-large\" src=\"https:\/\/voipstudio.com\/wp-content\/avada\/2025\/01\/pasted-image-0-1024x284.png\" alt=\"configurar stun voipstudio\" width=\"1024\" height=\"284\" \/><\/p>\n<ol start=\"3\">\n<li><span style=\"font-weight: 400;\"> Otros ajustes que se pueden realizar son:<\/span><\/li>\n<\/ol>\n<ul>\n<li><span style=\"font-weight: 400;\">Ajustar el protocolo de transporte de UDP a TCP.<\/span><\/li>\n<li><span style=\"font-weight: 400;\">Utilizar nuestro puerto SIP secundario 5566<\/span><\/li>\n<li><span style=\"font-weight: 400;\">Reducir el tiempo de reintento en el registro SIP<\/span><\/li>\n<li><span style=\"font-weight: 400;\">Configurar reglas de NAT est\u00e1ticas en el router para nuestro tel\u00e9fono IP.<\/span><\/li>\n<li><span style=\"font-weight: 400;\">Configurar una DMZ en el router y ubicar en la misma el tel\u00e9fono IP. En este caso es importante tomar medidas de seguridad adicionales en el control de acceso del terminal. No debemos dejarlo bajo ning\u00fan concepto con las credenciales por defecto.\u00a0\u00a0<\/span><\/li>\n<\/ul>\n<p><span style=\"font-weight: 400;\">Si eres cliente de <a href=\"https:\/\/voipstudio.com\/es\/\">VoIPstudio<\/a> y est\u00e1s sufriendo alg\u00fan problema de estos, no dudes en <strong>abrir un ticket de soporte<\/strong> con nosotros para ayudarte a diagnosticar y solventar el problema, tal como se muestra en la imagen:<\/span><\/p>\n<p><img decoding=\"async\" class=\"alignleft wp-image-40292 size-large\" src=\"https:\/\/voipstudio.com\/wp-content\/avada\/2025\/01\/pasted-image-0-1-1024x442.png\" alt=\"generar ticket soporte voipstudio\" width=\"1024\" height=\"442\" \/><br \/>\n<span style=\"font-weight: 400;\">\u00a0\u00a0<\/span><\/p>\n<p>&nbsp;<\/p>\n","protected":false},"excerpt":{"rendered":"<p>En ocasiones al recibir una llamada, esta se establece correctamente, y mientras estamos hablando con el otro interlocutor a los pocos segundos [&hellip;]<\/p>\n","protected":false},"author":4,"featured_media":37641,"comment_status":"closed","ping_status":"closed","sticky":false,"template":"","format":"standard","meta":{"inline_featured_image":false,"footnotes":""},"categories":[184],"tags":[210,508,193,506,495,113],"class_list":["post-26042","post","type-post","status-publish","format-standard","has-post-thumbnail","hentry","category-comunicaciones-empresariales","tag-ahorro-de-costes","tag-cloud-communications","tag-comunicaciones-en-la-nube","tag-cost-saving","tag-sip-and-voip-tech","tag-tecnologia-sip-y-voip"],"yoast_head":"<!-- This site is optimized with the Yoast SEO Premium plugin v27.3 (Yoast SEO v27.3) - https:\/\/yoast.com\/product\/yoast-seo-premium-wordpress\/ -->\n<title>Causas y Soluciones al Corte repentino de Llamadas en VoIP<\/title>\n<meta name=\"description\" content=\"Te contamos las causas y posibles soluciones para solucionar los cortes repentinos de llamadas en VoIP. 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