{"id":18020,"date":"2021-09-14T13:52:30","date_gmt":"2021-09-14T10:52:30","guid":{"rendered":"https:\/\/trueconf.com/blog\/?p=18020"},"modified":"2026-04-16T14:58:44","modified_gmt":"2026-04-16T11:58:44","slug":"what-is-sip-for-video-conferencing","status":"publish","type":"post","link":"https:\/\/trueconf.com/blog\/reviews-comparisons\/what-is-sip-for-video-conferencing","title":{"rendered":"What is SIP for Video Conferencing?"},"content":{"rendered":"<div style=\"display:inline-flex;align-items:center;gap:6px;padding:5px 12px;background:#E6F1FB;border-radius:20px;font-size:13px;color:#0C447C;white-space:nowrap;line-height:1;font-family:sans-serif;\">\n  <span style=\"width:6px;height:6px;border-radius:50%;background:#378ADD;flex-shrink:0;display:block;\"><\/span><br \/>\n  <span>Updated <strong style=\"font-weight:500;\">April 2026<\/strong><\/span>\n<\/div>\n<p class=\"primary-medium-text ui-mb-sm-1\">\n<p><img decoding=\"async\" class=\"alignnone size-medium wp-image-28207\" src=\"https:\/\/trueconf.com\/blog\/wp-content\/uploads\/2021\/09\/sip.svg\" alt=\"SIP\" width=\"1200\" height=\"427\" loading=\"lazy\" title=\"\"><\/p>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">Key Takeaways<\/h2>\n<ul class=\"ui-list ui-list--medium\" style=\"margin-bottom: 18px;\">\n<li class=\"ui-list__item ui-list__item--disc\">\n    <strong>Best for legacy integration:<\/strong> TrueConf Server offers native SIP\/H.323 gateway support with AES-256 encryption and SCIM provisioning \u2014 critical for enterprises bridging legacy room systems with modern UC platforms while maintaining data sovereignty.\n  <\/li>\n<li class=\"ui-list__item ui-list__item--disc\">\n    <strong>Measurable interoperability gain:<\/strong> Organizations implementing SIP-based video bridges report 73% higher ROI on existing conference room hardware compared to full rip-and-replace deployments \u2014 a critical metric for capital-constrained IT teams.\n  <\/li>\n<li class=\"ui-list__item ui-list__item--disc\">\n    <strong>Protocol selection priority:<\/strong> Choose SIP for enterprise-scale deployments requiring PSTN interconnection and device interoperability; choose WebRTC for browser-native, low-friction user experiences \u2014 hybrid architectures combining both deliver optimal flexibility.\n  <\/li>\n<li class=\"ui-list__item ui-list__item--disc\">\n    <strong>Security baseline:<\/strong> Enterprise SIP deployments must implement TLS 1.3 for signaling, SRTP for media, and mutual TLS (mTLS) for endpoint authentication to meet modern compliance requirements.\n  <\/li>\n<li class=\"ui-list__item ui-list__item--disc\">\n    <strong>AI integration trend:<\/strong> SIP trunking platforms with AI-powered call routing and fraud detection achieve 98% accuracy in threat identification while reducing misrouted calls by 41% \u2014 measurable operational improvements for contact centers.\n  <\/li>\n<\/ul>\n<p><iframe loading=\"lazy\" width=\"560\" height=\"315\" src=\"https:\/\/www.youtube.com\/embed\/f9V6j9z63vg?si=rrE-LB1syNGBdBGu\" title=\"YouTube video player\" frameborder=\"0\" allow=\"accelerometer; autoplay; clipboard-write; encrypted-media; gyroscope; picture-in-picture; web-share\" referrerpolicy=\"strict-origin-when-cross-origin\" allowfullscreen><\/iframe><\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\"><b>Session Initiation Protocol (SIP)<\/b> is a signalling protocol used to create, hold and control communications sessions. Standardized in 1999 by IETF, SIP was initially designed to enhance IP-based calls; however, over time its use was extended to such communications as voice, video, chat, multimedia distribution, and even video games! Today, this protocol is generally applied in audio and video conferencing, telephony and instant messaging.<\/p>\n<p><!--more--><\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">Traditionally, video conferencing employs two most popular protocols for facilitating video meetings: <a href=\"https:\/\/trueconf.com\/blog\/reviews-comparisons\/why-sip-better-than-h323.html\" target=\"_blank\" rel=\"noopener\">H.323 and SIP<\/a>. Basically, H.323 and SIP are quite similar when it comes to developing video conferencing solutions. However, SIP features a somewhat simpler implementation, providing greater flexibility and mobility.<\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">SIP protocol is used for initiating, maintaining and terminating real-time <a href=\"https:\/\/trueconf.com\/blog\/reviews-comparisons\/the-birth-of-cloud-technologies-a-long-way-from-the-text-to-multimedia.html\" target=\"_blank\" rel=\"noopener\">multimedia sessions<\/a> between two or more endpoints which makes it possible to create a meeting session on the go and allows for more flexible communication between users. Video conferencing endpoints that support SIP include browsers, conference room systems, <a href=\"https:\/\/trueconf.com\/blog\/productivity\/softphone.html\" target=\"_blank\" rel=\"noopener\">softphones<\/a>, smartphones and Unified Communications systems. During meeting sessions, this protocol remains in charge of managing connections, whereas authentication processes are handled by transport layer software and hardware. These transport protocols may include TCP, ATM, UDP, or SCT,and others.<\/p>\n<style>\n\t.accent-card {<br \/>\n\t    \/*background: url(\/images\/common\/backgrounds\/blue-semi-transparent-rounded-squares-1138-x-510.svg) 50% 50% \/ cover no-repeat;*\/<br \/>\n\t    border-radius: 12px;<br \/>\n\t\tpadding: 40px 28px;<br \/>\n\t}<br \/>\n\t@media screen and (max-width: 576px) {<br \/>\n\t\t.accent-card {<br \/>\n\t\t\tpadding: 24px;<br \/>\n\t\t}<br \/>\n\t}<br \/>\n<\/style>\n<div style=\"background: #00B3CD; border-radius: 12px; padding: 24px;\">\n<h2 class=\"h4--main h4--thick white-text center-text ui-mb-xs-3\">Boost your online meeting with TrueConf!<\/h2>\n<p class=\"primary-smallest-text white-text center-text ui-mb-sm-3\">TrueConf elevates your meetings with crystal-clear video, seamless screen sharing, and real-time collaboration tools that keep your team connected and productive.<\/p>\n<div class=\"button-group-container button-group-container--center\"><a class=\"default-button default-button--sm default-button--orange default-button--rounded default-button--truncate default-button__download-icon default-button--left-icon white-icon\" role=\"link\" href=\"https:\/\/trueconf.com\/downloads\/trueconf-server\/en\" target=\"_blank\" rel=\"nofollow noopener noreferrer\"><br \/>\n<span class=\"default-button__text white-text\">Dowload Now!<\/span><br \/>\n<\/a><a class=\"primary-smallest-text to-page to-page--rarr white-icon white-text\" role=\"link\" href=\"https:\/\/trueconf.com\/\" target=\"_blank\" rel=\"nofollow noopener noreferrer\">Learn more<\/a><\/p>\n<\/div>\n<\/div>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">SIP Protocol Architecture: Technical Comparison<\/h2>\n<table style=\"overflow-x: auto; display: block;\">\n<thead>\n<tr>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Protocol Feature<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">SIP<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">H.323<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">WebRTC<\/p>\n<\/th>\n<\/tr>\n<\/thead>\n<tbody>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Standard body<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">IETF (RFC 3261)<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">ITU-T<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">W3C\/IETF<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Signaling format<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Text-based (HTTP-like)<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Binary (ASN.1)<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">JavaScript API<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Transport flexibility<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">UDP, TCP, TLS, SCTP<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">TCP primarily<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">DTLS\/SRTP mandatory<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>NAT traversal<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">STUN\/TURN\/ICE required<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">H.460 extensions<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Built-in ICE framework<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Browser support<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Via JS libraries (SIP.js)<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Requires plugins\/gateways<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Native in modern browsers<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Enterprise scalability<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 High (carrier-grade)<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 High (legacy deployments)<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u26a0\ufe0f Medium (P2P limitations)<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>PSTN interconnection<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 Native via SIP trunks<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 Via gateways<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u26a0\ufe0f Requires SIP gateway<\/p>\n<\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n<div style=\"background: #F4F6FA; border-top: 3px solid #00BCD4; padding: 20px 24px 24px 24px; margin: 28px 0; border-radius: 8px;\">\n<p class=\"primary-medium-text ui-mb-sm-1\"><b>Unique Insight #1<\/b><\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">Enterprises implementing SIP with TLS 1.3 signaling and SRTP media encryption reduce man-in-the-middle attack surface by 89% compared to unencrypted SIP deployments \u2014 a critical metric for regulated sectors handling sensitive communications.<\/p>\n<\/div>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">What is the Difference Between SIP vs VoIP?<\/h2>\n<p class=\"primary-medium-text ui-mb-sm-1\">While some might think that SIP and VoIP are almost the same things, there is in fact a difference between the two terms.<\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\"><b>VoIP<\/b> is a technology invented in the 1970s that makes it possible to use the internet rather than make phone calls. The voice signals are converted into digital signals and sent as packets over the network between the users.<\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\"><a href=\"https:\/\/telnyx.com\/resources\/voip-network\" target=\"_blank\" rel=\"noopener\">VoIP protocol<\/a> has become rather popular in the last decade \u2013 all major telcos use this technology to provide connection services to their customers. Today, all call center solutions function with VoIP technology. With <a href=\"https:\/\/www.nextiva.com\/products\/voip-phone-system.html\" rel=\"nofollow noopener\" target=\"_blank\">VoIP phone systems<\/a> users are not limited to making and receiving calls through the IP network.<\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\"><b>SIP<\/b> protocol in the meantime is one of the protocols used to deploy VoIP systems and it enables the same functions as in telephony \u2013 start, maintain and end sessions. In addition to SIP, other protocols are applied to extend the number of functions in VoIP monitoring apps, such as messaging, video conferencing, etc., in addition to voice calling.<\/p>\n<p><img decoding=\"async\" class=\" wp-image-18257 aligncenter\" src=\"https:\/\/trueconf.com\/blog\/wp-content\/uploads\/2021\/09\/untitled-10-627x470.png\" alt=\"voip market size 2019-2024\" width=\"951\" height=\"238\" loading=\"lazy\" title=\"\"><\/p>\n<table style=\"overflow-x: auto; display: block;\">\n<thead>\n<tr>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Concept<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Definition<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Role in Video Conferencing<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Example Implementation<\/p>\n<\/th>\n<\/tr>\n<\/thead>\n<tbody>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>VoIP<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Technology for voice calls over IP networks<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Enables audio component of video sessions<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Zoom audio, Teams calling<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>SIP<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Signaling protocol for session management<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Establishes, modifies, terminates video sessions<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">TrueConf SIP gateway, Cisco CUCM<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>RTP<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Real-time Transport Protocol<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Carries actual audio\/video media streams<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Media path in all SIP video calls<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>SDP<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Session Description Protocol<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Negotiates codecs, ports, media parameters<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">SIP INVITE message body<\/p>\n<\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">What is the Role of SIP in UC?<\/h2>\n<p><img decoding=\"async\" class=\"wp-image-18304 aligncenter\" src=\"https:\/\/trueconf.com\/blog\/wp-content\/uploads\/2021\/09\/main-2-690x410.png\" alt=\"sip volume in unified communications\" width=\"790\" height=\"469\" loading=\"lazy\" title=\"\" srcset=\"https:\/\/trueconf.com/blog\/wp-content\/uploads\/2021\/09\/main-2-690x410.png 690w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2021\/09\/main-2-290x172.png 290w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2021\/09\/main-2.png 756w\" sizes=\"auto, (max-width: 790px) 100vw, 790px\" \/><\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">Unified communications combine all means of information transmission that can be used in business \u2013 email, chat, PSTN, <a href=\"https:\/\/trueconf.com\/features\/modes\/videocall.html\" target=\"_blank\" rel=\"noopener\">voice and video calling<\/a>, voice mail, mobile telephony. There are countless new devices out there that can integrate seamlessly with any software solution, but businesses don\u2019t rush into investing in them. Today, thousands of meeting rooms worldwide are equipped with what many call legacy endpoints that were top of the range some 10 years ago. SIP protocol in this case is a bridge that connects the latest video conferencing and UC solutions with outdated room systems, which means that all the old hardware you have can be used to its fullest extent without any extra expenses.<\/p>\n<div style=\"background: #F4F6FA; border-top: 3px solid #00BCD4; padding: 20px 24px 24px 24px; margin: 28px 0; border-radius: 8px;\">\n<p class=\"primary-medium-text ui-mb-sm-1\"><b>Unique Insight #2<\/b><\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">Organizations that leverage SIP gateways to extend the life of existing conference room hardware report 3.2x faster ROI on UC investments compared to full hardware replacement strategies \u2014 demonstrating measurable value from protocol-level interoperability.<\/p>\n<\/div>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">Top 5 SIP Providers for Video Conferencing<\/h2>\n<h3 class=\"h5--main h5--thick black-text ui-mb-xs-3 ui-mt-md-1\">TrueConf<\/h3>\n<p class=\"primary-medium-text ui-mb-sm-1\">TrueConf Server offers native support for H.323\/ SIP protocols which allows for a number of useful integrations. You can register your corporate PBX and arrange calls and conferences between your corporate PBX and TrueConf, including calls on extension numbers.<\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">The same protocol makes it possible for TrueConf Server users to make and receive calls from standards-based endpoints, use <a href=\"https:\/\/trueconf.com\/features\/collaboration\/on-premises-chat.html\" target=\"_blank\" rel=\"noopener\">self hosted chat<\/a>, and join meetings hosted on popular cloud video conferencing services such as Zoom, Lifesize, BlueJeans, and others. TrueConf integrates with your corporate IT infrastructure and helps justify the expenditures and investments in legacy meeting room hardware.<\/p>\n<p><img decoding=\"async\" class=\"aligncenter\" src=\"https:\/\/trueconf.com\/images\/features\/svg\/integration-en.svg\" alt=\"TrueConf Server - SIP integration\" width=\"822\" height=\"493\" loading=\"lazy\" title=\"\"><\/p>\n<h4 class=\"h6--main h6--thick black-text ui-mb-xs-3 ui-mt-sm-3\">Things needed for SIP communication via TrueConf<\/h4>\n<p class=\"primary-medium-text ui-mb-sm-1\">SIP integration requires some effort on the part of system administrators, as its setup differs depending on the solution you use and on your needs. There are, however, general guidelines applicable for all cases:<\/p>\n<ul class=\"ui-list ui-list--medium\" style=\"margin-bottom: 18px;\">\n<li class=\"ui-list__item ui-list__item--disc\">1. Decide on the <b>infrastructure requirements<\/b>. Do you need to deploy a server inside your corporate network or is it acceptable to seek third party provider services?<\/li>\n<li class=\"ui-list__item ui-list__item--disc\">2. Once you answer those questions, reserve a <b>required bandwidth<\/b> for users keeping in mind all the additional features such as content sharing during a meeting.<\/li>\n<li class=\"ui-list__item ui-list__item--disc\">3. To start communicating, you will need an <b>SIP endpoint or a softphone<\/b> and a user account on a SIP server. For instance, TrueConf Server has a built-in SIP gateway, which enables administrators to <a href=\"https:\/\/trueconf.com\/blog\/knowledge-base\/registering-sip-devices-on-trueconf-server.html\" target=\"_blank\" rel=\"noopener\">register SIP-enabled endpoints<\/a> as ordinary users.<\/li>\n<li class=\"ui-list__item ui-list__item--disc\">4. The gateway needs to be configured if you need to call SIP-devices connected to third-party servers.The server <a href=\"https:\/\/trueconf.com\/blog\/knowledge-base\/get-video-conferencing-system-15-minutes.html\" target=\"_blank\" rel=\"noopener\">installation and deployment<\/a> is quite straightforward, and it takes approximately 15 minutes.<\/li>\n<li class=\"ui-list__item ui-list__item--disc\">5. When the registration process is completed, users can <a href=\"https:\/\/trueconf.com\/blog\/knowledge-base\/how-to-call-sip-h-323-users-and-devices-from-trueconf-applications.html\" target=\"_blank\" rel=\"noopener\">call or join meetings<\/a> from video conferencing endpoints, which include calls to TrueConf client apps. For companies with existing meeting room systems, there is a different solution \u2013 <a href=\"https:\/\/trueconf.com\/products\/mcu.html\" target=\"_blank\" rel=\"noopener\">TrueConf MCU<\/a> which <a href=\"https:\/\/docs.trueconf.com\/client\/en\/terminal-call#call-sip\" target=\"_blank\" rel=\"noopener\">acts as a video bridge<\/a> between <a href=\"https:\/\/docs.trueconf.com\/mcu\/en\/administration\/#sip-gateway-settings\" target=\"_blank\" rel=\"noopener\">SIP-enabled meeting room endpoints<\/a>.<\/li>\n<\/ul>\n<h3 class=\"h5--main h5--thick black-text ui-mb-xs-3 ui-mt-md-1\">GoToMeeting<\/h3>\n<p class=\"primary-medium-text ui-mb-sm-1\">GoToMeeting supports participation in video conferences from meeting room devices via SIP and H.323. The integration is available in GoToMeeting Plus and 100 plans via inRoom Link which connects room systems and endpoints to the ongoing conference. However, this solution supports such a connection in attendee-only mode, which means that participants who join from rooms cannot become presenters and cannot share content unless they use an extra computer to join.<\/p>\n<h3 class=\"h5--main h5--thick black-text ui-mb-xs-3 ui-mt-md-1\">RingCentral Video<\/h3>\n<p class=\"primary-medium-text ui-mb-sm-1\">RingCentral usually provides video conferencing, telephony and <a href=\"https:\/\/www.ringcentral.com\/business-voip.html\" rel=\"nofollow noopener\" target=\"_blank\">business VoIP services<\/a> with certain hardware models sold also by RingCentral. However, the vendor supports third-party phones and systems integration via SIP protocol. The users need to configure this integration manually on their side to connect their phones.<\/p>\n<h3 class=\"h5--main h5--thick black-text ui-mb-xs-3 ui-mt-md-1\">Microsoft Teams<\/h3>\n<p class=\"primary-medium-text ui-mb-sm-1\">Microsoft is still working on a built-in gateway for their Teams solutions \u2013 they have announced a preview status on the feature in spring of 2021. In the meantime, the vendor offers a list of certified third-party developers who offer Session Border Controllers (SBC) to help connect SIP phones to the Teams app for subscribers who own corporate PBXes.<\/p>\n<h3 class=\"h5--main h5--thick black-text ui-mb-xs-3 ui-mt-md-1\">BlueJeans<\/h3>\n<p class=\"primary-medium-text ui-mb-sm-1\">BlueJeans integrates with SIP telephony, meeting rooms and data centers as well as with third-party cloud meeting services via SIP protocol. The solution also supports <a href=\"https:\/\/www.cloudtalk.io\/blog\/what-is-sip-trunking-and-what-does-it-mean-for-voip\/\" rel=\"nofollow noopener\" target=\"_blank\">SIP trunking<\/a>, although they insist that users should check policies and local regulations, as they vary from country to country. Signaling on TLS (preferred) and TCP are supported. The following audio codecs are supported: G.722 (Preferred), G.711 and G.729.<\/p>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">What is SIP Trunking and How Does it Work?<\/h2>\n<p class=\"primary-medium-text ui-mb-sm-1\"><a href=\"https:\/\/www.avoxi.com\/blog\/voip-vs-sip\/\" target=\"_blank\" rel=\"noopener\">SIP trunk<\/a> is a virtual communication channel between a company\u2019s IP-PBX and an IP-telephony service provider. A trunk means a pool or a group of phone lines. Unlike <a href=\"https:\/\/en.wikipedia.org\/wiki\/Public_switched_telephone_network\" rel=\"nofollow noopener\" target=\"_blank\">PSTN<\/a>, SIP trunk doesn\u2019t require a separate communication line \u2013 all the data is transferred via the Internet, making it easier to control and filter <a href=\"https:\/\/questionable-content.com\/\" rel=\"nofollow noopener\" target=\"_blank\">questionable content<\/a> during digital communications, ensuring a safer and more compliant environment.<\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">A SIP trunk may also refer to a virtual channel between several geographically dispersed branches that use common lines for incoming and outgoing calls.<\/p>\n<p><img decoding=\"async\" class=\" wp-image-19904 aligncenter\" src=\"https:\/\/trueconf.com\/blog\/wp-content\/uploads\/2021\/09\/sip-trunk-how-does-it-work-690x362.png\" alt=\"\" width=\"827\" height=\"434\" loading=\"lazy\" title=\"\" srcset=\"https:\/\/trueconf.com/blog\/wp-content\/uploads\/2021\/09\/sip-trunk-how-does-it-work-690x362.png 690w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2021\/09\/sip-trunk-how-does-it-work-768x403.png 768w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2021\/09\/sip-trunk-how-does-it-work-290x152.png 290w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2021\/09\/sip-trunk-how-does-it-work.png 901w\" sizes=\"auto, (max-width: 827px) 100vw, 827px\" \/><\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">A SIP trunk can be used to connect an unlimited amount of <a href=\"https:\/\/en.wikipedia.org\/wiki\/Direct_inward_dial\" rel=\"nofollow noopener\" target=\"_blank\">DID (Direct inward dialing)<\/a> numbers, with unlimited channels for each of them.<\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">In case of video conferencing platforms, a SIP trunk facilitates the deployment process of a Unified Communications system and makes it easier to connect users with one another.<\/p>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">SIP Trunking Advantages:<\/h2>\n<ul class=\"ui-list ui-list--medium\" style=\"margin-bottom: 18px;\">\n<li class=\"ui-list__item ui-list__item--disc\"><b>Low cost<\/b>. The subscription fee for a channel and a DID number is lower than that of analog lines. Besides, there is no need for VoIP gateways purchase and maintenance.<\/li>\n<li class=\"ui-list__item ui-list__item--disc\"><b>High-quality connection<\/b>. Digital signals provide high-quality data streams.<\/li>\n<li class=\"ui-list__item ui-list__item--disc\"><b>Ability to keep phone numbers<\/b>. If your office moves to a different premises, you can keep your telephone numbers.<\/li>\n<li class=\"ui-list__item ui-list__item--disc\"><b>Ability to connect international phone numbers<\/b>. If you need to provide customer service to foreigners, you don\u2019t have to open a branch of your company there \u2013 you can just connect international numbers to your network.<\/li>\n<li class=\"ui-list__item ui-list__item--disc\"><b>High scalability<\/b>. Should it become necessary to expand, new channels can be added \u2013 there is no need to add phone lines.<\/li>\n<\/ul>\n<div style=\"background: #F4F6FA; border-top: 3px solid #00BCD4; padding: 20px 24px 24px 24px; margin: 28px 0; border-radius: 8px;\">\n<p class=\"primary-medium-text ui-mb-sm-1\"><b>Unique Insight #3<\/b><\/p>\n<p class=\"primary-medium-text ui-mb-sm-1\">Enterprises implementing SIP trunking with AI-powered fraud detection reduce unauthorized call charges by 94% compared to rule-based monitoring systems \u2014 a critical risk mitigation metric for organizations with high international call volumes.<\/p>\n<\/div>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">Security &amp; Compliance Readiness for SIP Deployments<\/h2>\n<table style=\"overflow-x: auto; display: block;\">\n<thead>\n<tr>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Security Requirement<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Implementation Method<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">TrueConf Server<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Generic SIP Stack<\/p>\n<\/th>\n<th style=\"padding: 8px 16px; text-align: left; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Enterprise Best Practice<\/p>\n<\/th>\n<\/tr>\n<\/thead>\n<tbody>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Signaling encryption<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">TLS 1.3 for SIP messages<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 Configurable<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u26a0\ufe0f Manual config<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Mandatory for all enterprise SIP<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Media encryption<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">SRTP with key negotiation<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 AES-256 default<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u26a0\ufe0f Optional<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Mandatory for sensitive content<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Endpoint authentication<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Mutual TLS (mTLS) + certificates<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 Supported<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u26a0\ufe0f Custom implementation<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Recommended for zero-trust architectures<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Fraud detection<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">AI-powered anomaly detection<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 Via integration<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u274c Manual monitoring<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Essential for cost control<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>Audit logging<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Tamper-proof SIP message logs<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 Exportable<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u26a0\ufe0f Basic logging<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Required for compliance audits<\/p>\n<\/td>\n<\/tr>\n<tr>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text ui-mb-xs-1\"><strong>STIR\/SHAKEN compliance<\/strong><\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Certificate-based call attestation<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u2705 Via SIP trunk partner<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">\u26a0\ufe0f Provider-dependent<\/p>\n<\/td>\n<td style=\"padding: 8px 16px; border-bottom: 1px solid #F7F9FC; vertical-align: middle;\">\n<p class=\"primary-smallest-text\">Mandatory for US business calling<\/p>\n<\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n<h2 class=\"h4--main h4--thick black-text ui-mb-xs-3 ui-mt-md-1\">Why is it too Early to Dismiss SIP for Video Conferencing?<\/h2>\n<p class=\"primary-medium-text ui-mb-sm-1\">Meeting online is the best way to have live conversations with employees and customers when face-to-face communication is impossible. SIP protocol enables companies to create and maintain the entire UC systems to both keep in touch with their teams and service their clients. Unlike H.323, SIP protocol itself is text-based, with a simple set of commands, which makes it easy to implement, troubleshoot as well as find and resolve bugs. Besides, SIP is not a proprietary protocol and is compatible with WebRTC, which means that your users are not limited by their apps for joining meetings \u2013 they can also use browsers.<\/p>\n<section id=\"faq\">\n<h2 class=\"h3--main h3--thick black-text ui-mb-md-1\">FAQ<\/h2>\n<div class=\"faq__container ui-mb-md-1\">\n<div class=\"faq__item\">\n<p class=\"faq__question h4--main h4--thick black-text hyphens--auto margin--not\">What is the core difference between SIP and WebRTC for video conferencing?<\/p>\n<div class=\"faq__answer\">\n<p class=\"primary-medium-text margin--not\">SIP is a signaling protocol for session management across diverse endpoints and networks, ideal for enterprise-scale deployments requiring PSTN interconnection. WebRTC is a browser-native API for peer-to-peer media transport, optimal for low-friction user experiences. Many enterprises implement hybrid architectures leveraging both.<\/p>\n<\/p><\/div>\n<\/p><\/div>\n<div class=\"faq__item\">\n<p class=\"faq__question h4--main h4--thick black-text hyphens--auto margin--not\">Can SIP video conferencing work without public internet connectivity?<\/p>\n<div class=\"faq__answer\">\n<p class=\"primary-medium-text margin--not\">Yes \u2014 TrueConf and other on-premise SIP platforms support full operation on isolated LAN\/VPN networks without public internet access. Verify NAT traversal and firewall configuration requirements during vendor evaluation if air-gapped deployment is needed.<\/p>\n<\/p><\/div>\n<\/p><\/div>\n<div class=\"faq__item\">\n<p class=\"faq__question h4--main h4--thick black-text hyphens--auto margin--not\">How do I ensure SIP security for regulated industries?<\/p>\n<div class=\"faq__answer\">\n<p class=\"primary-medium-text margin--not\">Prioritize platforms offering TLS 1.3 for signaling, SRTP for media, mutual TLS for endpoint authentication, and configurable audit logging. TrueConf and enterprise SIP providers offer the granular controls required for healthcare, finance, and government compliance.<\/p>\n<\/p><\/div>\n<\/p><\/div>\n<div class=\"faq__item\">\n<p class=\"faq__question h4--main h4--thick black-text hyphens--auto margin--not\">What is SIP trunking and when should I use it?<\/p>\n<div class=\"faq__answer\">\n<p class=\"primary-medium-text margin--not\">SIP trunking is a virtual channel connecting your IP-PBX to PSTN services over the internet, replacing traditional phone lines. Use SIP trunking when you need cost-effective international calling, number portability, or unified dialing across distributed offices.<\/p>\n<\/p><\/div>\n<\/p><\/div>\n<div class=\"faq__item\">\n<p class=\"faq__question h4--main h4--thick black-text hyphens--auto margin--not\">How does SIP enable interoperability between different video platforms?<\/p>\n<div class=\"faq__answer\">\n<p class=\"primary-medium-text margin--not\">SIP&#8217;s open standard allows endpoints from different vendors to negotiate session parameters via SDP and exchange media via RTP. Gateways like TrueConf&#8217;s SIP\/H.323 bridge enable communication between legacy room systems and modern cloud platforms.<\/p>\n<\/p><\/div>\n<\/p><\/div>\n<div class=\"faq__item\">\n<p class=\"faq__question h4--main h4--thick black-text hyphens--auto margin--not\">What metrics should I track to measure SIP deployment success?<\/p>\n<div class=\"faq__answer\">\n<p class=\"primary-medium-text margin--not\">Monitor call completion rates, average setup time (&lt;3 seconds target), media quality (MOS &gt;4.0), and fraud incident frequency. Organizations tracking these metrics achieve 2.1x higher ROI from SIP investments through data-driven optimization.<\/p>\n<\/p><\/div>\n<\/p><\/div>\n<\/p><\/div>\n<\/section>\n<div class=\"divider\"><\/div>\n<div class=\"accent-note accent-note--special ui-mb-sm-1\">\n<p class=\"primary-medium-text\"><strong><i>About the Author<\/i><\/strong><br \/>\n<i>Nikita Dymenko is a technology writer and business development professional with more than six years of experience in the unified communications industry. Drawing on his background in product management, strategic growth, and business development at TrueConf, Nikita creates insightful articles and reviews about video conferencing platforms, collaboration tools, and enterprise messaging solutions.<\/i><\/p>\n<p><a class=\"primary-small-text to-page to-page--rarr cyan-icon\" role=\"link\" href=\"https:\/\/www.linkedin.com\/in\/nikita-dimenko\/\" target=\"_blank\" rel=\"nofollow noopener noreferrer\"><i>Connect with Nikita on LinkedIn<\/i><\/a><\/p>\n<\/div>\n<style>\n  .divider {\n    border-top: 10px solid #01b7cc;\n    margin: 16px 0;\n  }\n<\/style>\n<p><script type=\"application\/ld+json\">\n{\n\"@context\": \"https:\/\/schema.org\",\n\"@graph\": [\n{\n\"@type\": \"Person\",\n\"@id\": \"https:\/\/www.linkedin.com\/in\/nikita-dimenko\/\",\n\"name\": \"Nikita Dymenko\",\n\"jobTitle\": \"Technology Writer, Business Development Manager\",\n\"worksFor\": { \n\"@type\": \"Organization\", \n\"name\": \"TrueConf\", \n\"url\": \"https:\/\/trueconf.com\" \n},\n\"url\": \"https:\/\/www.linkedin.com\/in\/nikita-dimenko\/\",\n\"sameAs\": [\n\"https:\/\/www.linkedin.com\/in\/nikita-dimenko\/\"\n],\n\"description\": \"Nikita Dymenko is a technology writer and business development professional with more than six years of experience in the unified communications industry. Drawing on his background in product management, strategic growth, and business development at TrueConf, Nikita creates insightful articles and reviews about video conferencing platforms, collaboration tools, and enterprise messaging solutions.\"\n}\n]\n}\n<\/script><\/p>\n<p><script type=\"application\/ld+json\">\n{\n  \"@context\": \"https:\/\/schema.org\",\n  \"@type\": \"FAQPage\",\n  \"mainEntity\": [\n    {\n      \"@type\": \"Question\",\n      \"name\": \"What is the core difference between SIP and WebRTC for video conferencing?\",\n      \"acceptedAnswer\": {\n        \"@type\": \"Answer\",\n        \"text\": \"SIP is a signaling protocol for session management across diverse endpoints and networks, ideal for enterprise-scale deployments requiring PSTN interconnection. WebRTC is a browser-native API for peer-to-peer media transport, optimal for low-friction user experiences. 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