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Glossary

SIP (Session Initiation Protocol)

SIP (Session Initiation Protocol) is the standard IP telephony protocol, defined in RFC 3261. Phones, softphones, PBXs, and carriers use it to agree on a call: who is calling, whether the call is accepted, and when it ended. Almost every modern business phone system and contact center runs on SIP.

Why it mattersIf a system speaks SIP, you can connect any carrier and any IP phone without being tied to one vendor.

Also called: Session Initiation Protocol, SIP protocol, VoIP signaling

How it works

How SIP works

SIP carries only the control messages about a call. The voice itself travels separately, usually over RTP. The parties agree on codecs and media addresses through an SDP description carried inside SIP messages.

A basic call looks like this: the caller sends INVITE, the other side answers 180 Ringing, then 200 OK (picked up). The caller confirms with ACK and the conversation starts. Whoever hangs up first sends BYE.

  • REGISTER – a phone tells the server at which address to find it
  • INVITE – start a call
  • ACK – confirm the answer was received
  • BYE – end the call
  • CANCEL – cancel a call that has not been answered yet
  • OPTIONS – ask about a peer's capabilities or check that it is alive

Details

Who takes part

User agentWhoever makes and takes calls: an IP phone, a softphone, an IVR menu, or an AI voice agent.
RegistrarRemembers where each user can currently be reached.
Proxy serverPasses SIP requests on according to routing rules and filters out the rest; Kamailio is a well-known example.
SIP trunkA connection between a company's phone system and a carrier over SIP.

Practice

Ports and protection

By default SIP uses port 5060 (UDP or TCP), and port 5061 with TLS encryption. SIP servers exposed to the internet are scanned constantly by people trying to guess passwords and make calls at someone else's expense. So protection is needed from day one.

  • Encrypt signaling with TLS and voice with SRTP where the carrier supports it
  • Let only known IP addresses reach the server and use strong passwords
  • Put a SIP proxy in front of the telephony servers: it drops unknown requests and spreads the load
  • Block addresses automatically after several failed registration attempts
  • Watch call quality: latency, jitter (uneven delay), and packet loss

In CONTACT.WORK

How it works in CONTACT.WORK

All CONTACT.WORK telephony runs on SIP. Kamailio works as the SIP proxy, Asterisk and FreeSWITCH handle voice, IVR menus, and queues, and the softphone connects agents right from the browser. Carriers connect through SIP trunks, and regular IP phones work too. The setup is the same in the cloud and on the customer's servers.

More

Where to use it

Kamailio SIP proxy See how the SIP proxy protects telephony and routes calls Learn more Asterisk See how IVR menus and queues work Learn more Telephony Connect your carrier over SIP Learn more

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Related terms

SIP trunkWebRTCIP PBXSoftphoneCTI – computer telephony integration

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