When you call someone over the internet, two things have to happen. First, the two phones have to find each other and agree to talk. Then, the actual sound has to travel between them. SIP handles the first part only. Once the call connects, it steps aside and the voice goes a different way entirely, which turns out to explain most of the odd things that go wrong with internet phone calls.
What SIP is
SIP stands for Session Initiation Protocol. It's the language phones use to set up a call, change it, and end it.
Think of it as the usher rather than the conversation. SIP finds the person you're calling, asks if they're available, makes their phone ring, notices when they pick up, and tells both sides how to reach each other. Then it gets out of the way. When one of you hangs up, SIP comes back to close things down.
It was written in 2002, and it's the reason a desk phone from one manufacturer can call a mobile app from another through a carrier neither company works for. Everyone agreed to speak the same language.
SIP is built on the same ideas as the web. A phone sends a request, gets a numbered reply back, and everything is written in plain readable text. It's much less exotic than it sounds.
The one thing worth understanding
SIP does not carry your voice.
That surprises people, but it's deliberate. SIP arranges the call and hands over an address: essentially "send the audio here." The sound itself then travels in its own separate stream, over a different route, using a different system called RTP.
Two channels, doing two jobs. SIP is the introduction; RTP is the conversation.
This matters because it explains the most common internet-calling fault there is: the call connects, but nobody can hear anything. Or one person can hear and the other can't. That's not a broken call in the ordinary sense. The introduction worked perfectly and the conversation never found its way. Usually a router or firewall is blocking the audio path while letting the setup messages through.
If your VoIP calls ring and connect but have no sound, don't waste time on your phone settings. The problem is almost always the network between you and the other end.
What actually happens when you call someone
Every internet call follows roughly the same sequence. In plain terms:
| What's sent | Meaning |
|---|---|
| INVITE | I'd like to talk to you. Here's how to reach me. |
| 100 Trying | Got it, working on it. |
| 180 Ringing | Their phone is ringing now. |
| 200 OK | They answered. Here's how to reach them. |
| ACK | Understood. We're connected. |
| (audio) | The actual conversation, not SIP. |
| BYE | Someone hung up. |
| 200 OK | Confirmed. Call over. |
The whole exchange takes a fraction of a second, and it's why you hear a ringing tone at all: your phone plays that sound locally once the other end reports it's ringing.
A couple of things worth noticing. Either person can hang up, SIP treats both ends as equals once you're connected. And that block in the middle, where the audio lives, is the part SIP has nothing to do with.
Addresses that look like email
SIP identifies people with addresses that look deliberately familiar:
sip:[email protected]Phone numbers work the same way, just wrapped up in that format. When you dial an ordinary number from a VoIP phone, it becomes an address like this at the first stop, travels the internet that way, and converts back if it ends up on a traditional landline.
Your phone also checks in periodically to say where it currently is. That's why the same number reaches you whether your app is on hotel wifi or your desk phone is at the office, the network keeps a note of where to find you right now.
Numbers you might run into
If you ever see an error code on a VoIP system, these are the common ones:
- 486 Busy - the other line is engaged.
- 487 - the caller hung up before it was answered. Not an error, despite looking like one.
- 488 - the two ends couldn't agree on an audio format. A configuration problem, not a fault with either phone.
- 403 - the system refused the call, usually a login or permissions issue with the phone service.
- 503 - the service on the other end is unavailable.
Why SIP can't tell who's really calling
Here's the part that matters beyond troubleshooting. When a phone sends that first INVITE message, it includes the number to display to the person being called. SIP passes that along faithfully but it has no way of checking whether it's true.
That's not a bug that someone forgot to fix. SIP was designed for systems that cooperate, back when the only people who could originate calls were phone companies. Nothing in it was built to police identity.
This is the root of a lot of things you've probably experienced. It's why the number on your screen is displayed rather than verified, why showing a different number is a setting rather than a hack, and why the phone industry had to bolt on a separate verification system decades later to add the trust that was never there to begin with.
Is it secure?
Plainly: not by default. SIP messages travel as readable text unless encryption is switched on, and the audio needs its own separate encryption. Plenty of setups have one without the other.
If you run a phone system for a business, the practical version is: use the encrypted settings your provider offers, use strong passwords on every extension, and don't let a phone system sit directly exposed on the internet. Automated scanners find unprotected systems within hours, and the resulting fraudulent calls get billed to you.
!faq !faq-q What is SIP in simple terms? !faq-a It's the system phones use to set up an internet call. It finds the person you're calling, rings their phone, connects you when they answer, and ends the call when someone hangs up. It doesn't carry the sound itself.
!faq-q Is SIP the same thing as VoIP? !faq-a Not quite. VoIP is the general idea of making phone calls over the internet. SIP is the specific method used to set those calls up. Almost all VoIP uses SIP, which is why people mix the terms up.
!faq-q Why can I connect a call but not hear anything? !faq-a Because setting up the call and carrying the audio are two separate things travelling different paths. The setup worked; the audio didn't get through. It's nearly always a router or firewall blocking the sound, and it's the single most common VoIP problem there is.
!faq-q Do I need to know any of this to use a VoIP phone? !faq-a No. All of it happens automatically in the background. It's worth knowing only if you're troubleshooting a problem or setting up a phone system for a business.
!faq-q What is a SIP trunk? !faq-a It's the connection between a business phone system and its phone provider, delivered over the internet instead of physical phone lines. One connection handles many calls at once, and adding capacity is a settings change rather than an engineer visit.
!faq-q Can SIP tell if a caller is who they say they are? !faq-a No. It passes along whatever number the calling system provides, without checking it. That's why a separate verification framework called STIR/SHAKEN was added later, and why the number on your screen should be treated as a claim rather than proof. !faq-end
SIP is a small idea that turned out to be durable: get two phones talking, describe where the sound should go, then leave them to it. That restraint is why a protocol from 2002 still handles nearly every business call placed today, and why everything added since (encryption, caller verification, spam filtering) sits on top of it rather than replacing it.
