{"id":15386,"date":"2019-09-23T09:00:23","date_gmt":"2019-09-23T07:00:23","guid":{"rendered":"https:\/\/fonvirtual.com\/en\/?p=15386"},"modified":"2026-07-01T12:14:44","modified_gmt":"2026-07-01T10:14:44","slug":"what-are-sip-calls-and-how-does-it-work","status":"publish","type":null,"link":"https:\/\/fonvirtual.com\/en\/blog\/what-are-sip-calls-and-how-does-it-work\/","title":{"rendered":"What are SIP calls and how do they work"},"content":{"rendered":"\n<p class=\"wp-block-paragraph\">A <strong>SIP call<\/strong> is a voice or video call placed over the internet using the <strong>Session Initiation Protocol (SIP)<\/strong>, the signalling standard defined by the <a href=\"https:\/\/www.ietf.org\/\" target=\"_blank\" rel=\"noopener\">IETF<\/a> in <a href=\"https:\/\/datatracker.ietf.org\/doc\/html\/rfc3261\" target=\"_blank\" rel=\"noopener\">RFC 3261<\/a> (2002). SIP itself does not carry the voice \u2014 it negotiates the session (who is calling whom, which codecs to use, which ports to open) and then the audio is streamed separately using RTP. In 2026, the majority of UK and EU business telephony \u2014 cloud PBXs, contact centre platforms, VoIP handsets, softphones on laptops and mobiles \u2014 runs over SIP. This guide explains, with concrete 2026 examples, what a SIP call is, how it works step by step, and how it differs from WebRTC.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\">For a related practical topic, see our guide on <a href=\"https:\/\/fonvirtual.com\/en\/blog\/sip-number\/\">SIP numbers and how to get one<\/a>, and for a hands-on protocol comparison, our <a href=\"https:\/\/fonvirtual.com\/en\/blog\/switching-to-webrtc\/\">switching from SIP to WebRTC<\/a> analysis. If you need to understand the broader landscape, read our guide on the <a href=\"https:\/\/fonvirtual.com\/en\/blog\/best-virtual-pbx\/\">best virtual PBX<\/a> options.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\">What is a SIP call and how does the protocol work?<\/h2>\n\n\n\n<p class=\"wp-block-paragraph\">SIP is a <strong>text-based signalling protocol<\/strong>. Think of it as the digital equivalent of a switchboard operator: it knows who is registered, where they are reachable, and how to connect two endpoints. The voice itself travels over a separate channel (RTP, the Real-time Transport Protocol), but the call setup, ringing, hang-up, transfer, and conference features all use SIP messages.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\">A typical SIP call goes through six well-defined steps:<\/p>\n\n\n\n<ol class=\"wp-block-list\">\n<li><strong>Register<\/strong> \u2014 the SIP endpoint (IP phone, softphone, mobile app) sends a <code>REGISTER<\/code> message to a SIP server (the registrar), announcing its current IP address and username.<\/li>\n<li><strong>Invite<\/strong> \u2014 when the user dials a number, the client sends an <code>INVITE<\/code> request to the SIP proxy. The proxy looks up the destination in its location database.<\/li>\n<li><strong>Trying \/ Ringing<\/strong> \u2014 the proxy relays progress messages (<code>100 Trying<\/code>, <code>180 Ringing<\/code>) back to the caller while alerting the callee.<\/li>\n<li><strong>Negotiate codecs<\/strong> \u2014 once the callee accepts (<code>200 OK<\/code>), the two endpoints agree on which audio codec to use (G.711, G.722, Opus) via SDP (Session Description Protocol).<\/li>\n<li><strong>RTP media<\/strong> \u2014 voice packets start flowing over UDP, typically on even-numbered ports negotiated in the SDP exchange.<\/li>\n<li><strong>Hang up<\/strong> \u2014 either side sends a <code>BYE<\/code> request, and the proxy tears down the session.<\/li>\n<\/ol>\n\n\n\n<p class=\"wp-block-paragraph\">The full handshake, including the SDP offer\/answer and ICE candidates for NAT traversal, is detailed in our <a href=\"https:\/\/fonvirtual.com\/en\/blog\/switching-to-webrtc\/\">WebRTC vs SIP<\/a> comparison.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\">What do you need to make a SIP call in 2026?<\/h2>\n\n\n\n<p class=\"wp-block-paragraph\">Three things are essential, plus a few recommended extras:<\/p>\n\n\n\n<ul class=\"wp-block-list\">\n<li><strong>A SIP account<\/strong> (also called a SIP line, SIP trunk, or SIP number) provided by a VoIP operator or cloud PBX vendor. In the UK and EU, leading providers include Fonvirtual, 3CX, RingCentral, Genesys, BT Business, Gamma, and Vonage.<\/li>\n<li><strong>A SIP client<\/strong> (softphone) \u2014 an app on your laptop (Zoiper, Bria, Linphone, MicroSIP), on iOS\/Android, or a physical IP desk phone (Yealink, Polycom, Cisco, Gigaset).<\/li>\n<li><strong>A stable internet connection<\/strong> \u2014 about 100 kbps per concurrent HD call (G.722 or Opus codec). Fibre or cable Ethernet is preferred over Wi-Fi for call quality.<\/li>\n<li><em>Recommended:<\/em> a <strong>headset with microphone<\/strong>, and a <strong>QoS-aware router<\/strong> (802.1p or DSCP tagging) to prioritise voice traffic over general data.<\/li>\n<\/ul>\n\n\n\n<h2 class=\"wp-block-heading\">SIP calls vs WebRTC: what&#8217;s the difference?<\/h2>\n\n\n\n<p class=\"wp-block-paragraph\">WebRTC (Web Real-Time Communication) is a W3C\/IETF standard that has been built into every major browser since 2013. It delivers voice, video, and data natively from a web page, with end-to-end encryption (DTLS-SRTP) by default. SIP, by contrast, was designed for dedicated hardware and software clients and is the dominant protocol in business telephony infrastructure.<\/p>\n\n\n\n<figure class=\"wp-block-table is-style-stripes\"><table style=\"border-collapse:collapse;width:100%;\"><thead><tr><th style=\"border:1px solid #ccc;padding:8px;background:#f2f2f2;text-align:left;\">Aspect<\/th><th style=\"border:1px solid #ccc;padding:8px;background:#f2f2f2;text-align:left;\">SIP call<\/th><th style=\"border:1px solid #ccc;padding:8px;background:#f2f2f2;text-align:left;\">WebRTC call<\/th><\/tr><\/thead><tbody>\n<tr><td style=\"border:1px solid #ccc;padding:8px;\">Standard<\/td><td style=\"border:1px solid #ccc;padding:8px;\">RFC 3261 (IETF, 2002)<\/td><td style=\"border:1px solid #ccc;padding:8px;\">W3C + IETF (2013+)<\/td><\/tr>\n<tr><td style=\"border:1px solid #ccc;padding:8px;\">Media transport<\/td><td style=\"border:1px solid #ccc;padding:8px;\">RTP over UDP\/TCP<\/td><td style=\"border:1px solid #ccc;padding:8px;\">SRTP over UDP, DTLS-encrypted<\/td><\/tr>\n<tr><td style=\"border:1px solid #ccc;padding:8px;\">Encryption<\/td><td style=\"border:1px solid #ccc;padding:8px;\">Optional (SIPS\/TLS, SRTP)<\/td><td style=\"border:1px solid #ccc;padding:8px;\">Mandatory and native<\/td><\/tr>\n<tr><td style=\"border:1px solid #ccc;padding:8px;\">Typical device<\/td><td style=\"border:1px solid #ccc;padding:8px;\">IP phone, softphone, mobile app<\/td><td style=\"border:1px solid #ccc;padding:8px;\">Web browser, no install<\/td><\/tr>\n<tr><td style=\"border:1px solid #ccc;padding:8px;\">Typical use case<\/td><td style=\"border:1px solid #ccc;padding:8px;\">Business PBX, call centres, DIDs<\/td><td style=\"border:1px solid #ccc;padding:8px;\">Click-to-call, in-app voice, video<\/td><\/tr>\n<tr><td style=\"border:1px solid #ccc;padding:8px;\">Average latency<\/td><td style=\"border:1px solid #ccc;padding:8px;\">150\u2013300 ms<\/td><td style=\"border:1px solid #ccc;padding:8px;\">50\u2013150 ms<\/td><\/tr>\n<tr><td style=\"border:1px solid #ccc;padding:8px;\">UK PSTN switch-off (2025\u20132027)<\/td><td style=\"border:1px solid #ccc;padding:8px;\">Replaces ISDN directly<\/td><td style=\"border:1px solid #ccc;padding:8px;\">Co-exists; needs a SIP gateway to reach the PSTN<\/td><\/tr>\n<\/tbody><\/table><\/figure>\n\n\n\n<p class=\"wp-block-paragraph\">Bottom line: SIP is the protocol that powers most business phone systems today; WebRTC is the protocol that powers browser-based voice and video. Most modern cloud PBX platforms support both, routing SIP calls to the office and WebRTC calls to remote workers who only have a browser.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\">Common business use cases for SIP calls<\/h2>\n\n\n\n<ul class=\"wp-block-list\">\n<li><strong>Replacing ISDN lines<\/strong> \u2014 in the UK, the PSTN switch-off runs from 2025 to 2027 (Ofcom, 2023). SIP trunks are the direct replacement, with porting of existing geographic numbers (01\/02) typically completed in 1 working day.<\/li>\n<li><strong>Multi-site voice<\/strong> \u2014 a retailer with 20 branches uses SIP to merge all sites into a single dialling plan, with free internal calls and a centralised queue.<\/li>\n<li><strong>Remote and hybrid work<\/strong> \u2014 a sales rep installs a softphone on a laptop, registers to the company SIP server over VPN or WebSocket, and keeps their extension wherever they go.<\/li>\n<li><strong>Contact centre<\/strong> \u2014 SIP feeds the dialler, the IVR, and the agent screen. Call recording and quality monitoring run on top of the SIP signalling layer.<\/li>\n<li><strong>International presence<\/strong> \u2014 a London company can buy SIP numbers in Madrid, Paris, and Berlin (DIDs) and route them all to the same UK-based agents.<\/li>\n<\/ul>\n\n\n\n<h2 class=\"wp-block-heading\">Frequently asked questions about SIP calls<\/h2>\n\n\n\n<h3 class=\"wp-block-heading\">Are SIP calls free?<\/h3>\n\n\n\n<p class=\"wp-block-paragraph\">SIP calls <strong>between users of the same provider<\/strong> are typically free and unlimited. Calls to the public phone network (PSTN) are billed per minute or included in a bundle. In the UK, an unlimited SIP bundle to UK landlines and mobiles costs roughly \u00a310\u201325 per user per month in 2026, depending on the provider and contract length.<\/p>\n\n\n\n<h3 class=\"wp-block-heading\">Do SIP calls work abroad?<\/h3>\n\n\n\n<p class=\"wp-block-paragraph\">Yes, as long as you can reach your SIP server over the internet (Wi-Fi, 4G\/5G, or Ethernet). The cost of international calls follows your provider&#8217;s rate card. One caveat: the location transmitted to emergency services (999\/112 in the UK, 112 in the EU) is the address registered on the SIP line, not the caller&#8217;s actual GPS position. Travellers should keep a personal mobile for emergencies.<\/p>\n\n\n\n<h3 class=\"wp-block-heading\">How much bandwidth does a SIP call need?<\/h3>\n\n\n\n<p class=\"wp-block-paragraph\">Around <strong>100 kbps per call<\/strong> in HD (G.722 or Opus) and 64 kbps in standard quality (G.711). For 10 concurrent HD calls, plan for 1 Mbps dedicated to voice on top of general traffic. Enable QoS on your router to prioritise SIP and RTP packets over web browsing and downloads.<\/p>\n\n\n\n<h3 class=\"wp-block-heading\">Can SIP calls be recorded?<\/h3>\n\n\n\n<p class=\"wp-block-paragraph\">Yes. Most cloud PBX platforms (Fonvirtual, 3CX, RingCentral, Genesys) include call recording as a standard feature. In the UK, recording is permitted under UK GDPR if the caller is informed at the start of the call, the recording is stored securely, retained only as long as needed, and made available to the participants on request, in line with the <a href=\"https:\/\/ico.org.uk\/for-organisations\/uk-gdpr-guidance-and-resources\/\" target=\"_blank\" rel=\"noopener\">ICO<\/a> guidance.<\/p>\n\n\n\n<h3 class=\"wp-block-heading\">What is the difference between SIP and VoIP?<\/h3>\n\n\n\n<p class=\"wp-block-paragraph\"><strong>VoIP<\/strong> (Voice over IP) is the umbrella term for any voice transmission over IP networks. <strong>SIP<\/strong> is one specific signalling protocol used in VoIP \u2014 others include H.323, IAX2, and WebRTC. Since around 2010, SIP has become the de facto standard for business unified communications.<\/p>\n\n\n\n<h3 class=\"wp-block-heading\">Do I need a physical phone to make SIP calls?<\/h3>\n\n\n\n<p class=\"wp-block-paragraph\">No. A <strong>softphone<\/strong> (an app on your laptop, mobile, or browser) is enough. If you do want a desk phone, IP phones from Yealink, Polycom, Cisco, or Gigaset connect directly to a SIP server and behave like a traditional handset. For a deeper practical guide on calling extensions, see our <a href=\"https:\/\/fonvirtual.com\/en\/blog\/how-to-call-a-pbx-extension\/\">how to call a PBX extension<\/a> article.<\/p>\n\n\n\n<h3 class=\"wp-block-heading\">How secure are SIP calls?<\/h3>\n\n\n\n<p class=\"wp-block-paragraph\">Plain SIP is not encrypted and is vulnerable to eavesdropping, toll fraud, and denial-of-service attacks. To secure your calls, use <strong>SIPS<\/strong> (SIP over TLS) for signalling, <strong>SRTP<\/strong> for media, a <strong>SIP-aware firewall<\/strong>, and strong <strong>authentication<\/strong> on every endpoint. Modern providers enable these by default; if you self-host a PBX, configuration is your responsibility. Ofcom publishes a <a href=\"https:\/\/www.ofcom.org.uk\/phones-telecoms-and-internet\/advice-for-consumers\/scams\/phone-scams\" target=\"_blank\" rel=\"noopener\">guide to phone scams and call protection<\/a> that is useful background reading.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\" style=\"margin-top:30px;\">\u2014 <em>Fonvirtual editorial team \u00b7 Updated July 2026<\/em><\/p>\n\n","protected":false},"excerpt":{"rendered":"<p>SIP calls explained: what the Session Initiation Protocol is, how a SIP call works step by step (RFC 3261), and how it compares to WebRTC for business 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