SIP Drafts: PSTN Interworking

The following documents describe proposals for allowing SIP to carry ISUP messages and describe how SIP and PSTN signaling protocols can interwork. Note: The current drafts are draft-zimmerer-sip-bcp-t-00, draft-ietf-sip-isup, and draft-ietf-sip-isup-mime-04. Other drafts dealing with ISUP interworking are obsolete and should not be used. (These obsolete drafts include draft-zimmerer-mmusic-sip-bcp-t-00, draft-camarillo-mmusic-sip-isup-bcp-00, draft-ietf-mmusic-sip-isup-mime-00.) Implementors should be aware that is likely that even the above drafts will change significantly before being standardized. A summary of efforts in this area.

A test specification is at Telenor Signalling System No. 7 Norwegian national interconnect - Test descriptions for Version 2 interface additional tests: ISDN-SIP end-to-end.

RFC 2976: The SIP INFO Method
Steven R. Donovan.
July 2000. (approved August 2000 as Proposed Standard)
This document proposes an extension to the Session Initiation Protocol (SIP). This extension adds the INFO method to the SIP protocol. The intent of the INFO method is to allow for the carrying of session related control information that is generated during a session. One example of such session control information is ISUP and ISDN signaling messages used to control telephony call services. This and other example uses of the INFO method may be standardized in the future.

Mapping of ISUP Overlap Signalling to SIP
G. Camarillo, A. Roach, J. Peterson, L. Ong
May 2001
This document describes a way to map ISUP overlap signalling to SIP.

Signaled Digits in SIP

R. Mahy
February 2001.
This document demonstrates a way for interested SIP User Agents which are not a party to the media of a call or session, to receive SIP event notifications when signaled digits, or other specific telephony-related events are detected. This is useful for a variety of applications that monitor calls for a specific event (e.g.: a long pound, special sequence of digits, or a fax signal) and--only then--take an active role in the monitored calls.

Mapping of ISUP parameters to SIP headers in SIP-T

Jon Peterson and Lyndon Ong
February 2001.
This document defines procedures within SIP-T for translation of ISUP call context to SIP call context and vice versa to allow ISUP calls to pass through SIP networks while preserving feature transparency.

Providing Emergency Call Services for SIP-based Internet Telephony
H. Schulzrinne
March 2001.
If Internet Telephony is to offer a full replacement for traditional telephone services, it needs to provide emergency call services. In the United States, emergency calls are known as 911 services, based on the number dialed. This note desccribes some options for providing enhanced emergency service, i.e., emergency calls that allow emergency response centers to determine the address where the caller is located. This is made more difficult by the temporary nature of IP addresses, the large number of ISPs and their lack of legal responsibility for emergency services and the ability of many Internet terminals to be connected to the Internet at different locations. This note explores some of the requirements and design choices.

E.164 Resolution in SIP
B. Campbell
November 2000.
This document describes how SIP may use the DNS to resolve services associated with E.164 numbers, as specified in draft-ietf-enum-e164-dns.

SIP-IN Interworking Protocol Architecture and Procedures
F. Haerens
February 2001
This draft gives a first input on the SIP-IN Interworking Protocol Architecture and Procedures for further discussion into the IETF as part of the SIP-IN Interworking (SIN design team).

SIP enabled IN Services - an implementation report
V. Gurbani
November 2000.
SIP is an excellent vehicle for the converged network services of the future; of that there is no doubt. However, even in the near term, SIP is an equally powerful solution to bridge the PSTN and VoIP networks by its application to the IN (Intelligent Network) services domain. This Internet Draft details our experiences of applying SIP to the said domain. We use a SIP call controller to execute IN services by mapping IN call model states to those of the SIP protocol state machine. [2] uses the notion of call model integration, an example of which is to use the IN call model as a canonical call model to map the protocol states of IP (Internet Protocol) based call controllers (SIP, H.323,...) to those of the IN call model.

Accessing IN services from SIP networks
V. Gurbani
November 2000.
In Internet telephony, the call control functions of a traditional circuit switch are replaced by a IP-based call controller that must provide features normally provided by the traditional switch, including operating as a SSP for IN features. A traditional switch is armed with an IN call model that provides it a means to reach out and make service decisions based on intelligence stored elsewhere. Internet call controllers, by contrast, do not have an IN call model. Furthermore, since there are many Internet call models with varying number of states than the IN call model, there has to be a mapping from the IN call model states to the equivalent states of the Internet call model if existing services are to be accessed transparently. To leverage the existing IN services from the Internet domain, this draft proposes a mapping from the states of the IN call model to the states of SIP, an Internet call signaling protocol.

Ringback tones in SIP-Based Telephony
Adam Roach
November 2000.
This document describes a mechanism by which an appropriate ringback tone may be played to the calling party when the called party's device is alerting. It is written specifcally to address the case where the Session Initiation Protocol (SIP) is used to initiate voice-over-IP calls. It also lists ringback characteristics for several countries.

SIP T.38 Call Flow Examples And Best Current Practice
J. Mule.
March 2001.
The Session Initiation Protocol allows the establishment of real- time Internet fax communications as defined by the ITU-T T.38 recommendation. This document attempts to clarify the options available to Internet telephony gateway vendors to handle real-time fax calls using SIP.

SIP for Telephones (SIP-T): Context and Architectures
Aparna Vemuri and Jon Peterson
March 2001.
SIP-T (earlier referred to as the SIP-BCP-T) is a mechanism that uses SIP to facilitate the interconnection of the PSTN with IP. This document explains the context and the architectures in which SIP-T may be used. This document has to be studied in conjunction with the existing SIP-T (referred to in some older documents as SIP-BCP-T) literature.

MIME media types for ISUP and QSIG Objects
E. Zimmerer, J. Peterson, A. Vemuri, L. Ong, M. Watson, M. Zonoun
September 2000.
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 [1]. 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 between legacy systems.

Interworking between SIP and INAP
H. Schulzrinne, L. Slutsman, I. Faynberg.
July 2000.
The goal of this document is to identify a new IETF work item. The document defines the term 'soft switch' as a mechanism by which PSTN Intelligent Network (IN) service control can be accessed by VoIP gateways and associated SIP servers. The document mechanism for interworking of the Session Initiation Protocol (SIP) and Intelligent Network Application Part Protocol (INAP).

SIP/IN Interworking
D. Lebovits.
July 2000
The goal of this document is to identify and propose a new work item for the IETF. This document describes the PSTN Intelligent Network (IN) Architecture support of Session Initiation Protocol (SIP) networks. The concept of the 'Soft-SSF' is introduced which acts as an overlay between the IP telephony call control and the Intelligent Network layer provided by the IN Service Switching Function (SSF) and the IN Service Control Function (SCF). This 'Soft-SSF' provides the necessary mapping between the SIP protocol state machine and the IN Basic Call State Model (BCSM). Also introduced is the Call Manager Function (CMF) which acts as a Mediation Node. The CMF entity is responsible for passing service related information to and from IN service layer, namely the SCF, and managing the service control relationship. As such, the CMF may contain a SSF-like functionality or subset thereof, to model the pre and post conditions that are required to interact with an SCF. The document specifically deals with the proposed standardized interfaces between the functional entities identified in the IN Network with associated functional entities represented in the SIP network. A mapping of parameters of the SIP protocol to the Intelligent Network Application Part Protocol(INAP) may be required for the support of the SIP Proxy Server call control protocol, states and events. Thereby enabling the Mediation Node (CMF) to model a SIP Proxy server. It is the proposal of this document to define the Mediation Node(CMF)to Soft SSF interface as a work item in the IETF as it is presently not a subject for standardization.

SIP INFO Method for Event Reporting
V. Bharatia, E. Cave, B. Culpepper. April 2000.
This document describes the use of the SIP INFO Method for communicating mid-call events in SIP sessions. Two new MIME types are described, according to the rules defined in RFC 2046 [2], for use in the INFO message. These media types can be used within a SIP INFO message to request, and report, event detection between SIP network entities. Emphasis is placed on DTMF signaling to communicate user indications when using SIP between a Media Gateway Controller (MGC) and a SIP application.

ISUP to SIP Mapping
Gonzalo Camarillo, Adam Roach, Jon Peterson, Lyndon Ong.
May 2001.
This document describes a way to perform the mapping between two signaling protocols: the Session Initiation Protocol (SIP) and the ISDN User Part (ISUP) of SS7.

Sample Uses of SIP INFO with Varying Reliability Needs
J. Kuthan. October 1999.

ISUP parameters expected in SIP messages
A. Roach

SIP Best Current Practice for Telephony Interworking (replaces "SIP+")
E. Zimmerer, A. Vemuri, V. Nadkarni, B. Morgan, G. Camarillo. October 1999.

Best Current Practice for ISUP to SIP mapping
G. Camarillo, August 1999. Obsolete.

L. Ong, F. Audet, M. Zonoun, E. Zimmerer, A. Vemuri. October 1999.

SIP PSTN Interworking Umbrella 'Require:' Header
A. Roach

Provisional SIP Responses with Media
A. Roach

A Functional Description of a SIP-PSTN Gateway
Steve Donovan and Matthew Cannon

A Proposal for Internet Call Waiting Service using SIP: An Implementation Report
A. Brusilovsky, E. Gausmann, V. Gurbani, A. Jain

SDP Extensions for Fax over IP Using T.38
Elin Wedlund, Wenyu Jiang, Henning Schulzrinne

Last updated by Henning Schulzrinne