SIP Architecture Overview
SIP (Session Initiation Protocol, RFC 3261) is a signaling protocol for establishing, modifying, and terminating multimedia sessions — primarily VoIP calls. SIP handles the *setup* of a call; the actual audio/video is carried separately over RTP (Real-time Transport Protocol).
Key SIP components:
- User Agent Client (UAC): initiates requests (the caller)
- User Agent Server (UAS): responds to requests (the callee)
- SIP Proxy: routes requests between UAC and UAS
- SIP Registrar: maps
user@domainto a current IP/port - SIP URI:
sip:[email protected]orsips:[email protected](TLS)
INVITE Transaction
A call begins with an INVITE request from the caller. The INVITE body contains an SDP (Session Description Protocol) offer describing the caller's audio capabilities:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP alice-pc.example.com:5060;branch=z9hG4bK776asdhds
From: Alice <sip:[email protected]>;tag=1928301774
To: Bob <sip:[email protected]>
Call-ID: [email protected]
CSeq: 314159 INVITE
Content-Type: application/sdp
v=0
o=alice 2890844526 2890844526 IN IP4 alice-pc.example.com
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000Provisional Responses
Before the callee answers, the SIP proxy and UAS send provisional responses (1xx):
SIP/2.0 100 Trying100 Trying — sent by the proxy to stop the UAC from retransmitting the INVITE. Does not indicate Bob's phone is ringing yet.
SIP/2.0 180 Ringing180 Ringing — Bob's phone is alerting. The caller hears a ringback tone.
SIP/2.0 183 Session Progress183 Session Progress — used with early media (e.g., playing an announcement before the call is answered).
200 OK and ACK
When Bob answers, his UA sends 200 OK with an SDP *answer* (his audio capabilities):
SIP/2.0 200 OK
From: Alice <sip:[email protected]>;tag=1928301774
To: Bob <sip:[email protected]>;tag=a6c85cf
Content-Type: application/sdp
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000Alice's UA sends an ACK to confirm receipt of the 200 OK. The ACK completes the three-way handshake:
ACK sip:[email protected] SIP/2.0After the ACK, RTP audio begins flowing directly between Alice and Bob (peer-to-peer, bypassing the SIP proxy).
BYE and Call Termination
Either party can end the call by sending BYE:
BYE sip:[email protected] SIP/2.0
From: Bob <sip:[email protected]>;tag=a6c85cf
To: Alice <sip:[email protected]>;tag=1928301774
Call-ID: [email protected]The other side responds with 200 OK, and both UAs release the RTP session.
SIP Registration Flow
Before receiving calls, a UA registers its current location with a Registrar so the proxy knows where to route INVITEs:
REGISTER sip:example.com SIP/2.0
From: Alice <sip:[email protected]>
To: Alice <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Expires: 3600The registrar responds with 200 OK listing active bindings. Re-registration must occur before the Expires interval ends.
SIP Error Scenarios
| Code | Meaning | Common Cause |
|---|---|---|
| `486 Busy Here` | Callee is busy | Another call in progress |
| `487 Request Terminated` | INVITE was cancelled | Caller hung up before answer |
| `404 Not Found` | User not found | Not registered |
| `408 Request Timeout` | No response | Network issue |
| `503 Service Unavailable` | Proxy unavailable | Overload or downtime |
487 Request Terminated is not an error — it is the normal flow when a caller hangs up before the callee answers. Seeing large numbers of 487s in logs is normal for interactive calling applications.