RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures

Publisher

RFC Editor / IETF

URL

https://www.rfc-editor.org/rfc/rfc3372.html

Published

2002-09-01

Updated

unknown

Accessed

2026-09-24

HTTP status

200

License

IETF Trust Legal Provisions; no-redistribution; short excerpts and locators only

Source notes citing this source

• RFC 3372 (SIP-T: Context and Architectures) is Best Current Practice BCP 63, published September 2002. in context
• SIP-T identifies two gateway functions: encapsulation, which bridges PSTN signalling across a SIP network in multipart MIME bodies, and translation, which maps ISUP messages and parameters to SIP messages and headers. in context
• RFC 3372 (September 2002, BCP 63) defines SIP-T as a set of mechanisms for interfacing traditional telephone signalling with SIP, not a new protocol. in context
• In SIP-T, full ISUP messages are carried end to end by encapsulating them in multipart MIME bodies of SIP messages, while key ISUP parameters are also translated into SIP headers for routing. in context
• RFC 3372 recommends the SIP INFO method for carrying mid-call ISUP signalling that does not change SIP call state. in context
• RFC 3372 sets three SIP-T requirements: transparency of ISUP information across a SIP segment, routability of SIP requests using header information, and transfer of mid-call signalling. in context

Cite this source record

APA

WarmTransfer. (2002, September 1). RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures. WarmTransfer. https://warmtransfer.net/knowledge/sources/rfc-3372-sip-t

BibTeX

@misc{warmtransfer-rfc-3372-sip-t,
  title  = {RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures},
  author = {{WarmTransfer}},
  year   = {2002},
  url    = {https://warmtransfer.net/knowledge/sources/rfc-3372-sip-t},
  note   = {RFC Editor / IETF, accessed 2026-09-24}
}