SIP INVITE — Session Initiation Protocol INVITE message
Nedir SIP INVITE?
Bu terim şu anda yalnızca İngilizce olarak sunulmaktadır.
A SIP INVITE is the initial Session Initiation Protocol request sent to set up a VoIP call, containing caller identity, destination details, and an SDP payload describing supported audio codecs and media ports. It triggers the standard three-way handshake — INVITE, 200 OK, ACK — that establishes bidirectional RTP media streams between endpoints, PBX systems, softswitches, and carrier session border controllers.
More detail
The SIP INVITE packet carries essential signaling headers including From, To, Call-ID, CSeq, and Contact, alongside an SDP (Session Description Protocol) message body that negotiates audio/video codecs like Opus or G.711 and RTP media ports.
During call setup, the receiving softswitch or gateway returns provisional responses such as 100 Trying and 180 Ringing before returning 200 OK when the call is answered, which the caller acknowledges with an ACK before audio flows over RTP.
A re-INVITE occurs when session parameters need to change on an established call — such as placing a call on hold, performing a warm transfer, enabling call recording, or renegotiating codecs between transcoding hops.
Sık sorulan sorular
- What is the SIP INVITE three-way handshake?
- The three-way handshake consists of an initial INVITE from the caller, a 200 OK response from the recipient when the call is answered, and an ACK from the caller confirming receipt of the response, finalizing session setup before voice media streams.
- What is the difference between a SIP INVITE and a re-INVITE?
- A SIP INVITE creates a brand-new call dialog and negotiates initial media parameters, whereas a re-INVITE is sent within an existing active dialog to modify attributes such as placing a party on hold, altering media IP addresses, or switching codecs.
- How does a SIP INVITE relate to RTP media streams?
- A SIP INVITE handles the signaling path only — negotiating endpoints, codecs, and port numbers via SDP — while the actual live conversation audio travels separately across direct peer-to-peer or relay RTP (Real-time Transport Protocol) streams once the handshake completes.
See also
Build it on Orbit
Voice, messaging, email, video, and AI agents on one platform and one pay-as-you-go bill. Start free — no credit card required.