The short answer
SIP (Session Initiation Protocol) sets calls up, changes them, and ends them. It does not carry your voice, RTP does that on a separate stream. Browser-to-browser calls use WebRTC and need no SIP at all. When a browser call has to reach a real phone number, SIP is what bridges it into the traditional phone network. It is the plumbing you never see until something breaks.
What SIP actually stands for
Session Initiation Protocol. Published by the Internet Engineering Task Force in 1999 as RFC 2543, and revised in 2002 as RFC 3261, SIP defines how internet-connected devices negotiate and establish real-time communication sessions. Its authors deliberately modeled it on HTTP and SMTP, the protocols behind web browsing and email, which is why SIP messages are human-readable text instead of dense binary data.
A SIP message looks roughly like this:
INVITE sip:+12025551234@carrier.com SIP/2.0
From: <sip:caller@freecallme.com>
To: <sip:+12025551234@carrier.com>
Content-Type: application/sdpThat is an invitation to start a call with a specific number. The carrier receives it, works out where that number lives, and starts the connection on the other end. Simple enough to read, powerful enough to route billions of calls a day.
What SIP does, and what it does not
SIP handles three things.
Call setup. When you dial a number, a SIP INVITE goes from your client to a SIP server (a proxy or registrar), which routes it toward the destination. The destination accepts with a '200 OK'. Your client acknowledges with an 'ACK'. The call is connected.
Call modification. During a call, SIP can update the session: switching from audio to video, adding a participant, or putting a call on hold. Each is a new SIP transaction inside the same ongoing session.
Call termination. When either party hangs up, a SIP 'BYE' is sent, the other side acknowledges it, and the session ends. Everything is cleaned up.
What SIP does not do is carry your voice. Once it has confirmed both sides are ready to talk, the audio travels over RTP (Real-time Transport Protocol), a separate stream that runs independently of the SIP signaling. That separation is intentional: the two streams can be optimized on their own, and a hiccup in signaling does not necessarily interrupt audio already in progress.
SIP vs VoIP: what is the difference?
People use "SIP" and "VoIP" interchangeably in casual conversation, but they are not the same thing.
VoIP (Voice over Internet Protocol) is the broad category: any way of sending voice as data over an IP network. It describes what is being done. SIP is a specific protocol many VoIP systems use to manage call signaling. It describes how calls are set up. SIP is one way to do VoIP; there are others, including H.323, WebRTC with its own signaling, and proprietary approaches like early Skype.
The shorthand: VoIP is the goal, SIP is one of the main tools for reaching it. If you want the wider picture, start with what VoIP is and how it works.
SIP vs WebRTC: the key distinction
If you have read the WebRTC explainer, you might wonder how the two relate. WebRTC handles real-time media entirely within the browser. It has its own signaling layer, usually over WebSockets, and sends media peer-to-peer using DTLS-SRTP. You do not need SIP to make a WebRTC call.
SIP is what connects calls to the traditional phone network. When a browser call needs to reach a real phone number, the browser's WebRTC audio gets handed to a SIP-based infrastructure that terminates the call on the phone network.
In practice, FreeCallMe's browser-to-browser calls use WebRTC throughout. A FreeCallMe dialer call to a real phone number uses WebRTC from your browser to a licensed carrier, and SIP is what carries the call onward from that carrier into the traditional phone network. The two technologies meet at the gateway.
See it in action
Make a browser call, or dial a real number and let WebRTC and SIP do the work behind the scenes.
Call any phone number in 220+ countries from your browser. Your first $0.25 is free, then pay-as-you-go by the minute.
What SIP trunking is
"SIP trunking" is a common term in business telephony that often causes confusion. A trunk, in old phone terminology, is a shared connection between two switching systems, the bundle of lines linking a company's internal phone system to the outside world.
SIP trunking replaces those physical lines with an internet connection. Instead of renting copper circuits from a carrier to connect your office phone system to the outside world, a SIP trunk connects your system to a VoIP provider over the internet, and calls travel as data.
For many businesses, switching to SIP trunking cuts phone costs substantially (figures of 40 to 60% are commonly cited) and removes the hardware needed for traditional PRI (Primary Rate Interface) lines. It also makes capacity elastic: adding lines is a configuration change, not a physical installation.
Why SIP messages look the way they do
SIP was designed to be extensible, able to handle not just phone calls but any session that needs two parties to negotiate. That has made it useful well beyond telephony: it shows up in video conferencing, instant messaging, and some IoT setups where devices need to establish direct connections.
The text-based format, while less efficient than a binary protocol, makes SIP easy to debug. An engineer chasing a failed call can read the SIP trace directly and see exactly which message failed and why. That was a deliberate trade-off the authors made in favor of interoperability and being able to troubleshoot.
Frequently asked questions
- Is SIP the same as VoIP?
- No. VoIP is the broad category, sending voice as data over an IP network. SIP is one specific protocol many VoIP systems use to set up and tear down calls. VoIP is the goal; SIP is one of the main tools for reaching it.
- What is the difference between SIP and RTP?
- SIP is signaling: it sets up, changes, and ends the call. RTP (Real-time Transport Protocol) carries the actual audio once the call is connected. They run as separate streams, so the voice and the control messages can be optimized independently.
- Do I need SIP to make a browser call?
- Not for a browser-to-browser call. That runs on WebRTC, which has its own signaling and sends media peer-to-peer. SIP enters the picture when a browser call has to reach a real phone number, where it connects the call into the traditional phone network.
- What is a SIP trunk?
- A SIP trunk replaces the physical phone lines that once connected a company's phone system to the outside world with an internet connection to a VoIP provider. Calls travel as data, capacity becomes a configuration change rather than a physical install, and it commonly cuts phone costs.
- What port does SIP use?
- SIP typically uses port 5060 for unencrypted signaling and port 5061 for TLS-encrypted signaling. The media stream (RTP) uses a separate range of ports negotiated during call setup.
The short version
SIP sets calls up. RTP carries the audio. WebRTC handles everything for browser-to-browser calls. When a call goes from a browser to a real phone, SIP connects those two worlds at the gateway. STIR/SHAKEN attestation rides inside SIP and has been required of major US carriers since 2021, which is why a call from FreeCallMe's dialer arrives with a verified Level A caller ID rather than a self-asserted one. It is not something most people need to think about, until something goes wrong, and then knowing what SIP does explains a lot about why internet calling works the way it does.
Put the protocol to work
Share a link for a free browser call, or dial a real phone number from your browser. The plumbing sorts itself out.
Call any phone number in 220+ countries from your browser. Your first $0.25 is free, then pay-as-you-go by the minute.