📡 You’re Invited… to the Session! (What Even is SIP?)

“SIP sets up the game. RTP plays it.”

🧩 So... What is SIP?

SIP (Session Initiation Protocol) is the VoIP equivalent of the game master — it sets the rules, gets the players into the arena, and then steps back once the game (the voice conversation) begins.

In plain terms:

SIP handles call setup, ringing, answering, and tear-down. It does not carry your voice — that’s RTP’s job.

Think of SIP like sending someone a party invite. It tells them:

Who’s calling
Where the call should happen
How they can reach you
And whether they want to accept it or ghost you

☎️ The Call Flow (Simplified)

Here’s how a basic SIP call between two softphones works:

User A            SIP Server            User B
  |                   |                   |
  |--- INVITE ------->|                   |
  |                   |--- INVITE ------->|
  |                   |<-- 180 RINGING ---|
  |<-- 180 RINGING ---|                   |
  |                   |<-- 200 OK --------|
  |<-- 200 OK --------|                   |
  |--- ACK ---------->|                   |
  |                   |--- ACK ---------->|
  |                   |                   |
  |<==== RTP Media Stream (Voice) ======>|
  |                   |                   |
  |--- BYE ---------->|                   |
  |                   |--- BYE ---------->|
  |                   |<-- 200 OK --------|
  |<-- 200 OK --------|                   |

🧠 Key SIP Messages

Message	What it Means
INVITE	“Hey, want to talk?”
180 RINGING	“They’re being notified”
200 OK	“They picked up!”
ACK	“Cool, let’s connect”
BYE	“Gotta go, bye!”

🎯 SIP in Action — What You See in Logs

If you’ve ever looked at SIP logs (Wireshark, anyone?), you’ll see something like:

INVITE sip:bob@example.com SIP/2.0
From: "Alice" sip:alice@example.net
To: sip:bob@example.com
Call-ID: 789abc123@example.net
CSeq: 1 INVITE
Contact: sip:alice@192.168.1.10:5060
Content-Type: application/sdp
Content-Length: 151

Notice that last part? Content-Type: application/sdp 👀

Yup — SIP includes something called SDP inside it. It’s like the fine print in your call invitation — it lists what codecs you'll use, what port to send media to, and more.

🧩 SIP ≠ Voice

SIP doesn’t carry your voice.

Instead, once the SIP call is accepted, RTP (Real-time Transport Protocol) takes over to actually move the media (voice, video, etc.) between endpoints.

So:

SIP = The call setup (the RSVP)
RTP = The actual voice (the conversation)

🔁 What Happens When You Hang Up?

A typical SIP BYE exchange:

User A --> BYE --> SIP Server --> BYE --> User B
User B --> 200 OK --> SIP Server --> 200 OK --> User A

Simple. Fast. And very polite.

🎯 TL;DR

SIP is a signaling protocol, not a media protocol
It helps set up, modify, and end VoIP calls
It uses messages like INVITE, RINGING, ACK, and BYE
It wraps media setup details using SDP (coming next)
After the call setup, RTP moves the actual voice packets

🧠 Up Next in SIP GAMES:

“What’s in the Envelope? Unpacking SDP”

SDP (Session Description Protocol) tells your devices what media codecs to use, how to send the audio, and more — it’s like the secret note inside your SIP envelope.

Stay tuned as we dive into how a voice call actually finds its voice.

🕹 Follow @sip_games for more VoIP magic — minus the RFC migraines.

SIP GAMES @sip_games