A SIP call is a voice or video call placed over the internet using the Session Initiation Protocol (SIP), the signalling standard defined by the IETF in RFC 3261 (2002). SIP itself does not carry the voice — 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 — cloud PBXs, contact centre platforms, VoIP handsets, softphones on laptops and mobiles — 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.

For a related practical topic, see our guide on SIP numbers and how to get one, and for a hands-on protocol comparison, our switching from SIP to WebRTC analysis. If you need to understand the broader landscape, read our guide on the best virtual PBX options.

What is a SIP call and how does the protocol work?

SIP is a text-based signalling protocol. 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.

A typical SIP call goes through six well-defined steps:

1. Register — the SIP endpoint (IP phone, softphone, mobile app) sends a REGISTER message to a SIP server (the registrar), announcing its current IP address and username.
2. Invite — when the user dials a number, the client sends an INVITE request to the SIP proxy. The proxy looks up the destination in its location database.
3. Trying / Ringing — the proxy relays progress messages (100 Trying, 180 Ringing) back to the caller while alerting the callee.
4. Negotiate codecs — once the callee accepts (200 OK), the two endpoints agree on which audio codec to use (G.711, G.722, Opus) via SDP (Session Description Protocol).
5. RTP media — voice packets start flowing over UDP, typically on even-numbered ports negotiated in the SDP exchange.
6. Hang up — either side sends a BYE request, and the proxy tears down the session.

The full handshake, including the SDP offer/answer and ICE candidates for NAT traversal, is detailed in our WebRTC vs SIP comparison.

What do you need to make a SIP call in 2026?

Three things are essential, plus a few recommended extras:

• A SIP account (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.
• A SIP client (softphone) — an app on your laptop (Zoiper, Bria, Linphone, MicroSIP), on iOS/Android, or a physical IP desk phone (Yealink, Polycom, Cisco, Gigaset).
• A stable internet connection — about 100 kbps per concurrent HD call (G.722 or Opus codec). Fibre or cable Ethernet is preferred over Wi-Fi for call quality.
• Recommended: a headset with microphone, and a QoS-aware router (802.1p or DSCP tagging) to prioritise voice traffic over general data.

SIP calls vs WebRTC: what’s the difference?

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.

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.

Common business use cases for SIP calls

• Replacing ISDN lines — 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.
• Multi-site voice — a retailer with 20 branches uses SIP to merge all sites into a single dialling plan, with free internal calls and a centralised queue.
• Remote and hybrid work — 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.
• Contact centre — SIP feeds the dialler, the IVR, and the agent screen. Call recording and quality monitoring run on top of the SIP signalling layer.
• International presence — a London company can buy SIP numbers in Madrid, Paris, and Berlin (DIDs) and route them all to the same UK-based agents.

Frequently asked questions about SIP calls

Are SIP calls free?

SIP calls between users of the same provider 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 £10–25 per user per month in 2026, depending on the provider and contract length.

Do SIP calls work abroad?

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’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’s actual GPS position. Travellers should keep a personal mobile for emergencies.

How much bandwidth does a SIP call need?

Around 100 kbps per call 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.

Can SIP calls be recorded?

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 ICO guidance.

What is the difference between SIP and VoIP?

VoIP (Voice over IP) is the umbrella term for any voice transmission over IP networks. SIP is one specific signalling protocol used in VoIP — others include H.323, IAX2, and WebRTC. Since around 2010, SIP has become the de facto standard for business unified communications.

Do I need a physical phone to make SIP calls?

No. A softphone (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 how to call a PBX extension article.

How secure are SIP calls?

Plain SIP is not encrypted and is vulnerable to eavesdropping, toll fraud, and denial-of-service attacks. To secure your calls, use SIPS (SIP over TLS) for signalling, SRTP for media, a SIP-aware firewall, and strong authentication on every endpoint. Modern providers enable these by default; if you self-host a PBX, configuration is your responsibility. Ofcom publishes a guide to phone scams and call protection that is useful background reading.

— Fonvirtual editorial team · Updated July 2026