Internet Drafts and RFCs: Interworking

We recommend you make yourself familiar with how this archive operates before you start using it.

H.323 rfc3372 rfc3578 rfc3204 draft-ietf-sipping-sipt rfc3398 draft-schulzrinne-sin draft-gurbani-iptel-sip-to-in draft-gurbani-sin draft-elwell-sipping-qsig2sip rfc2719 rfc2960 rfc2833 rfc2458 rfc2848 draft-zhao-iptel-gwloc-slp

rfc3372.txt	Session Initiation Protocol for Telephones (SIP-T): Context and Architectures
Author(s)	A. Vemuri, J. Peterson
Organization	ietf
State	best current practice
Size	49893 bytes
also	bcp63.txt
Abstract	The popularity of gateways that interwork between the PSTN (Public Switched Telephone Network) and SIP networks has motivated the publication of a set of common practices that can assure consistent behavior across implementations. This document taxonomizes the uses of PSTN-SIP gateways, provides uses cases, and identifies mechanisms necessary for interworking. The mechanisms detail how SIP provides for both 'encapsulation' (bridging the PSTN signaling across a SIP network) and 'translation' (gatewaying).

rfc3204.txt	MIME media types for ISUP and QSIG Objects
Author(s)	E. Zimmerer, J. Peterson, A. Vemuri, L. Ong, F. Audet, M. Watson, M. Zonoun
Organization	ietf
State	proposed standard
Size	19712 bytes
updated by	rfc3459.txt
Abstract	This document describes MIME types for application/ISUP and application/QSIG objects for use in SIP applications, according to the rules defined in RFC 2048. These types can be used to identify ISUP and QSIG objects within a SIP message such as INVITE or INFO, as might be implemented when using SIP in an environment where part of the call involves interworking to the PSTN.

draft-ietf-sipping-sipt-04.txt

"SIP for Telephones (SIP-T): Context and Architectures", A. Vemuri, J. Peterson, 07/05/2002,
The popularity of gateways that interwork between the PSTN (PublicSwitched Telephone Network) and SIP networks has motivated thepublication of a set ofcommon practices that can assure consistentbehavior across implementations. This document taxonomizes the usesof PSTN-SIP gateways, provides uses cases, and identifies mechanismsnecessary for interworking. The mechanisms detail how SIP providesfor both 'encapsulation' (carriage of PSTN signaling across a SIPnetwork) and 'translation' (protocol mapping).

wdiff comparison with previous version

rfc3398.txt Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping

Author(s) G. Camarillo, A. B. Roach, J. Peterson, L. Ong

Organization ietf

State proposed standard

Size 166207 bytes

Abstract This document describes a way to perform the mapping between two signaling protocols: the Session Initiation Protocol (SIP) and the Integrated Services Digital Network (ISDN) User Part (ISUP) of Signaling System No. 7 (SS7). This mechanism might be implemented when using SIP in an environment where part of the call involves interworking with the Public Switched Telephone Network (PSTN).

draft-schulzrinne-sin-02.txt

wdiff comparison with previous version

Note that this document is expired.

draft-gurbani-iptel-sip-to-in-06.txt

wdiff comparison with previous version

draft-gurbani-sin-02.txt Interworking SIP and Intelligent Network (IN) Applications

Author(s) Vijay Gurbani, F Haerens, Vidhi Rastogi

Organization ietf

State unknown

Date 2002-06-21

Size 62152 bytes

Abstract Public Switched Telephone Network (PSTN) services such as 800 number routing (freephone), time-and-day routing, credit-card calling, virtual private network (mapping a private network number into a public number) are realized by the Intelligent Network (IN). This draft addresses means to support existing IN services from Session Initiation Protocol (SIP) endpoints for an IP-host-to-phone call. The call request is originated on a SIP endpoint, but the services to the call are provided by the data and procedures resident in the PSTN/IN. To provide IN services in a transparent manner to SIP endpoints, this draft describes the mechanism for interworking SIP and Intelligent Network Application Part (INAP). Table of Contents 1 INTRODUCTION.................................................. 3 2 ACCESS TO IN-SERVICES FROM A SIP ENTITY....................... 4 3 ADDITIONAL SIN CONSIDERATIONS................................. 7 3.1 The concept of state in SIP.............................. 7 3.2 Relationship between SCP and a SIN-enabled SIP entity.... 8 3.3 SIP REGISTER and IN services............................. 8 3.4 Support of announcements and mid-call signaling.......... 8 4 THE SIN ARCHITECTURE.......................................... 9 4.1 Definitions.............................................. 9 4.2 IN Service control based on the SIN approach.............10 5 MAPPING OF THE SIP STATE MACHINE TO THE IN STATE MODEL........11 5.1 Mapping SIP protocol state machine to O_BCSM.............12 5.2 Mapping SIP protocol state machine to T_BCSM.............17 6 EXAMPLE CALL FLOWS............................................22 7 SECURITY CONSIDERATIONS.......................................23 Appendix A.......................................................23 Normative References.............................................24 Informative References...........................................24 Acknowledgments..................................................25 Changes from previous drafts.....................................25 Author's addresses...............................................26

draft-elwell-sipping-qsig2sip-03.txt Interworking between SIP and QSIG

Author(s) John Elwell

Organization ietf

State unknown

Date 2002-10-24

Size 160749 bytes

Abstract This document specifies interworking between the Session Initiation Protocol (SIP) and QSIG within corporate telecommunication networks (also known as enterprise networks). SIP is an Internet application- layer control (signalling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include, in particular, telephone calls. QSIG is a signalling protocol for creating, modifying and terminating circuit-switched calls, in particular telephone calls, within Private Integrated Services Networks (PISNs). QSIG is specified in a number of ECMA Standards and published also as ISO/IEC standards. As the support of telephony within corporate networks evolves from circuit-switched technology to Internet technology, the two technologies will co-exist in many networks for a period, perhaps several years. Therefore there is a need to be able to establish, modify and terminate sessions involving a participant in the SIP Elwell et alia Expires - April 2003 [Page 1] Interworking between SIP and QSIG October 2002 network and a participant in the QSIG network. Such calls are supported by gateways that perform interworking between SIP and QSIG.

wdiff comparison with previous version

rfc2719.txt Framework Architecture for Signaling Transport

Author(s) L. Ong, I. Rytina, M. Garcia, H. Schwarzbauer, L. Coene, H. Lin, I. Juhasz, M. Holdrege, C. Sharp

Organization ietf

State informational

Size 48646 bytes

Abstract This document defines an architecture framework and functional requirements for transport of signaling information over IP. The framework describes relationships between functional and physical entities exchanging signaling information, such as Signaling Gateways and Media Gateway Controllers. It identifies interfaces where signaling transport may be used and the functional and performance requirements that apply from existing Switched Circuit Network (SCN) signaling protocols.

rfc2960.txt Stream Control Transmission Protocol

Author(s) R. Stewart, Q. Xie, K. Morneault, C. Sharp, H. Schwarzbauer, T. Taylor, I. Rytina, M. Kalla, L. Zhang, V. Paxson

Organization ietf

State proposed standard

Size 297757 bytes

updated by rfc3309.txt

Abstract This document describes the Stream Control Transmission Protocol (SCTP). SCTP is designed to transport PSTN signaling messages over IP networks, but is capable of broader applications. SCTP is a reliable transport protocol operating on top of a connectionless packet network such as IP. It offers the following services to its users: -- acknowledged error-free non-duplicated transfer of user data, -- data fragmentation to conform to discovered path MTU size, -- sequenced delivery of user messages within multiple streams, with an option for order-of-arrival delivery of individual user messages, -- optional bundling of multiple user messages into a single SCTP packet, and -- network-level fault tolerance through supporting of multi- homing at either or both ends of an association. The design of SCTP includes appropriate congestion avoidance behavior and resistance to flooding and masquerade attacks.
FYI: SCTP bake-off info may be found at ftp://standards.nortelnetworks.com/sigtran-test/ . A free reference implementation is available at http://www.sctp.chicago.il.us .

rfc2833.txt RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

Author(s) H. Schulzrinne, S. Petrack

Organization ietf

State proposed standard

Size 68786 bytes

Abstract This memo describes how to carry dual-tone multifrequency (DTMF) signaling, other tone signals and telephony events in RTP packets.

rfc2458.txt Toward the PSTN/Internet Inter-Networking--Pre-PINT Implementations

Author(s) H. Lu, M. Krishnaswamy, L. Conroy, S. Bellovin, F. Burg, A. DeSimone, K. Tewani, P. Davidson, H. Schulzrinne, K. Vishwanathan

Organization ietf

State informational

Size 139154 bytes

Abstract This document contains the information relevant to the development of the inter-networking interfaces underway in the Public Switched Telephone Network (PSTN)/Internet Inter-Networking (PINT) Working Group. It addresses technologies, architectures, and several (but by no means all) existing pre-PINT implementations of the arrangements through which Internet applications can request and enrich PSTN telecommunications services. The common denominator of the enriched services (a.k.a. PINT services) is that they combine the Internet and PSTN services in such a way that the Internet is used for non-voice interactions, while the voice (and fax) are carried entirely over the PSTN. One key observation is that the pre-PINT implementations, being developed independently, do not inter-operate. It is a task of the PINT Working Group to define the inter-networking interfaces that will support inter-operation of the future implementations of PINT services.

rfc2848.txt The PINT Service Protocol: Extensions to SIP and SDP for IP Access to Telephone Call Services

Author(s) S. Petrack, L. Conroy

Organization ietf

State proposed standard

Size 168851 bytes

Abstract This document contains the specification of the PINT Service Protocol 1.0, which defines a protocol for invoking certain telephone services from an IP network. These services include placing basic calls, sending and receiving faxes, and receiving content over the telephone. The protocol is specified as a set of enhancements and additions to the SIP 2.0 and SDP protocols.

draft-zhao-iptel-gwloc-slp-05.txt Locating IP-to-Public Switched Telephone Network (PSTN) Telephony Gateways via SLP

Author(s) Henning Schulzrinne, Weibin Zhao

Organization ietf

State unknown

Date 2002-08-29

Size 22741 bytes

Abstract This document describes how to use the Service Location Protocol (SLP) to locate Internet telephony gateways. It defines the "service:iptel-gw" template for the Internet telephony gateway service, and discusses the different usage scenarios and the applicability of SLP for the Internet telephony gateway location.

wdiff comparison with previous version

1998-2002, maintained by Jiri Kuthan.
Last Update: May 14, 2002

rfc3398.txt	Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping
Author(s)	G. Camarillo, A. B. Roach, J. Peterson, L. Ong
Organization	ietf
State	proposed standard
Size	166207 bytes
Abstract	This document describes a way to perform the mapping between two signaling protocols: the Session Initiation Protocol (SIP) and the Integrated Services Digital Network (ISDN) User Part (ISUP) of Signaling System No. 7 (SS7). This mechanism might be implemented when using SIP in an environment where part of the call involves interworking with the Public Switched Telephone Network (PSTN).

draft-gurbani-sin-02.txt	Interworking SIP and Intelligent Network (IN) Applications
Author(s)	Vijay Gurbani, F Haerens, Vidhi Rastogi
Organization	ietf
State	unknown
Date	2002-06-21
Size	62152 bytes
Abstract	Public Switched Telephone Network (PSTN) services such as 800 number routing (freephone), time-and-day routing, credit-card calling, virtual private network (mapping a private network number into a public number) are realized by the Intelligent Network (IN). This draft addresses means to support existing IN services from Session Initiation Protocol (SIP) endpoints for an IP-host-to-phone call. The call request is originated on a SIP endpoint, but the services to the call are provided by the data and procedures resident in the PSTN/IN. To provide IN services in a transparent manner to SIP endpoints, this draft describes the mechanism for interworking SIP and Intelligent Network Application Part (INAP). Table of Contents 1 INTRODUCTION.................................................. 3 2 ACCESS TO IN-SERVICES FROM A SIP ENTITY....................... 4 3 ADDITIONAL SIN CONSIDERATIONS................................. 7 3.1 The concept of state in SIP.............................. 7 3.2 Relationship between SCP and a SIN-enabled SIP entity.... 8 3.3 SIP REGISTER and IN services............................. 8 3.4 Support of announcements and mid-call signaling.......... 8 4 THE SIN ARCHITECTURE.......................................... 9 4.1 Definitions.............................................. 9 4.2 IN Service control based on the SIN approach.............10 5 MAPPING OF THE SIP STATE MACHINE TO THE IN STATE MODEL........11 5.1 Mapping SIP protocol state machine to O_BCSM.............12 5.2 Mapping SIP protocol state machine to T_BCSM.............17 6 EXAMPLE CALL FLOWS............................................22 7 SECURITY CONSIDERATIONS.......................................23 Appendix A.......................................................23 Normative References.............................................24 Informative References...........................................24 Acknowledgments..................................................25 Changes from previous drafts.....................................25 Author's addresses...............................................26

draft-elwell-sipping-qsig2sip-03.txt	Interworking between SIP and QSIG
Author(s)	John Elwell
Organization	ietf
State	unknown
Date	2002-10-24
Size	160749 bytes
Abstract	This document specifies interworking between the Session Initiation Protocol (SIP) and QSIG within corporate telecommunication networks (also known as enterprise networks). SIP is an Internet application- layer control (signalling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include, in particular, telephone calls. QSIG is a signalling protocol for creating, modifying and terminating circuit-switched calls, in particular telephone calls, within Private Integrated Services Networks (PISNs). QSIG is specified in a number of ECMA Standards and published also as ISO/IEC standards. As the support of telephony within corporate networks evolves from circuit-switched technology to Internet technology, the two technologies will co-exist in many networks for a period, perhaps several years. Therefore there is a need to be able to establish, modify and terminate sessions involving a participant in the SIP Elwell et alia Expires - April 2003 [Page 1] Interworking between SIP and QSIG October 2002 network and a participant in the QSIG network. Such calls are supported by gateways that perform interworking between SIP and QSIG.

rfc2719.txt	Framework Architecture for Signaling Transport
Author(s)	L. Ong, I. Rytina, M. Garcia, H. Schwarzbauer, L. Coene, H. Lin, I. Juhasz, M. Holdrege, C. Sharp
Organization	ietf
State	informational
Size	48646 bytes
Abstract	This document defines an architecture framework and functional requirements for transport of signaling information over IP. The framework describes relationships between functional and physical entities exchanging signaling information, such as Signaling Gateways and Media Gateway Controllers. It identifies interfaces where signaling transport may be used and the functional and performance requirements that apply from existing Switched Circuit Network (SCN) signaling protocols.

rfc2960.txt	Stream Control Transmission Protocol
Author(s)	R. Stewart, Q. Xie, K. Morneault, C. Sharp, H. Schwarzbauer, T. Taylor, I. Rytina, M. Kalla, L. Zhang, V. Paxson
Organization	ietf
State	proposed standard
Size	297757 bytes
updated by	rfc3309.txt
Abstract	This document describes the Stream Control Transmission Protocol (SCTP). SCTP is designed to transport PSTN signaling messages over IP networks, but is capable of broader applications. SCTP is a reliable transport protocol operating on top of a connectionless packet network such as IP. It offers the following services to its users: -- acknowledged error-free non-duplicated transfer of user data, -- data fragmentation to conform to discovered path MTU size, -- sequenced delivery of user messages within multiple streams, with an option for order-of-arrival delivery of individual user messages, -- optional bundling of multiple user messages into a single SCTP packet, and -- network-level fault tolerance through supporting of multi- homing at either or both ends of an association. The design of SCTP includes appropriate congestion avoidance behavior and resistance to flooding and masquerade attacks.

rfc2833.txt	RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
Author(s)	H. Schulzrinne, S. Petrack
Organization	ietf
State	proposed standard
Size	68786 bytes
Abstract	This memo describes how to carry dual-tone multifrequency (DTMF) signaling, other tone signals and telephony events in RTP packets.

rfc2458.txt	Toward the PSTN/Internet Inter-Networking--Pre-PINT Implementations
Author(s)	H. Lu, M. Krishnaswamy, L. Conroy, S. Bellovin, F. Burg, A. DeSimone, K. Tewani, P. Davidson, H. Schulzrinne, K. Vishwanathan
Organization	ietf
State	informational
Size	139154 bytes
Abstract	This document contains the information relevant to the development of the inter-networking interfaces underway in the Public Switched Telephone Network (PSTN)/Internet Inter-Networking (PINT) Working Group. It addresses technologies, architectures, and several (but by no means all) existing pre-PINT implementations of the arrangements through which Internet applications can request and enrich PSTN telecommunications services. The common denominator of the enriched services (a.k.a. PINT services) is that they combine the Internet and PSTN services in such a way that the Internet is used for non-voice interactions, while the voice (and fax) are carried entirely over the PSTN. One key observation is that the pre-PINT implementations, being developed independently, do not inter-operate. It is a task of the PINT Working Group to define the inter-networking interfaces that will support inter-operation of the future implementations of PINT services.

rfc2848.txt	The PINT Service Protocol: Extensions to SIP and SDP for IP Access to Telephone Call Services
Author(s)	S. Petrack, L. Conroy
Organization	ietf
State	proposed standard
Size	168851 bytes
Abstract	This document contains the specification of the PINT Service Protocol 1.0, which defines a protocol for invoking certain telephone services from an IP network. These services include placing basic calls, sending and receiving faxes, and receiving content over the telephone. The protocol is specified as a set of enhancements and additions to the SIP 2.0 and SDP protocols.

draft-zhao-iptel-gwloc-slp-05.txt	Locating IP-to-Public Switched Telephone Network (PSTN) Telephony Gateways via SLP
Author(s)	Henning Schulzrinne, Weibin Zhao
Organization	ietf
State	unknown
Date	2002-08-29
Size	22741 bytes
Abstract	This document describes how to use the Service Location Protocol (SLP) to locate Internet telephony gateways. It defines the "service:iptel-gw" template for the Internet telephony gateway service, and discusses the different usage scenarios and the applicability of SLP for the Internet telephony gateway location.