SIP - Session Initiation Protocol

Cisco Aug 31, 2025

peer-to-peer
created by the IETF
Based on existing protocols (HTTP, SMTP)
Used text-based ASCII communication
loop detection via SIP header information
3 times less overhead than H.323 for call setup
H.323 maintains call state information for every call, overhead on the gateway

SIP signaling purposes

determine location
determine media capabilities (codecs supported)
determine availability
establish session
manage termination/transfer

SIP Clients are in two parts

User Agent Client (UAC) - starts calls
User Agent Server (UAS) - receives requests from calls

SIP Server Roles

Redirect Server - takes incoming calls and does redirection
Registrar Server - endpoints register with server, updates location server
Location Server - holds database of where endpoints are located
Proxy Server - performs all the SIP server roles

SIP Call Setup

Delayed Offer

UAC sends an INVITE to UAS
UAS sends 100 TRYING
UAS sends 180 RINGING
UAS sends 200 OK (SDP:Media Offer)
UAC sends ACK (SDP:Answer)
RTP is setup between endpoints

Early Offer - negotiates codec before establishing the call, SDP is embed in either the invite or ringing message.

sip configuration

user agent configuration

Gateway will register all of their dial-peers with a registrar server

sip-ua
  register ipv4:10.0.1.10
  authentication username NAME password PASSWORD

Dial-peer configuration

!! sip server per dial peer

dial-peer voice 1 voip
  session target ipv4:10.0.1.10
  sessions protocol sip

!!global sip server

sip-ua
  sip-server ipv4:10.0.1.10
	
dial-peer voice 1 voip
  session target sip-server
  sessions protocol sip

Global commands

voice service voip
  sip
    session transport <UDP/TCP> !! default is UDP port 5060, can also set under a dial-peer
    bind <control/media/all> source-inferface