Internet Engineering Task Force                     Gonzalo Camarillo
Internet draft                                             Adam Roach
<draft-ietf-sipping-overlap-00.txt>                          Ericsson
January 2002
Expires: July 2002                                       Jon Peterson

                                                           Lyndon Ong

               Mapping of ISUP Overlap Signalling to SIP

Status of this Memo

   This document is an Internet-Draft and is in full conformance with
      all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that
   other groups may also distribute working documents as Internet-
   Drafts. Internet-Drafts are draft documents valid for a maximum of
   six months and may be updated, replaced, or obsoleted by other
   documents at any time. It is inappropriate to use Internet- Drafts
   as reference material or to cite them other than as "work in

   The list of current Internet-Drafts can be accessed at
   The list of Internet-Draft Shadow Directories can be accessed at


   This document describes a way to map ISUP overlap signalling to SIP.

Camarillo/Roach/Peterson/Ong                                         1

               Mapping of ISUP Overlap Signalling to SIP


   1   Introduction.................................................2
   2   Overlap signalling in SIP....................................2
   3   ISUP to SIP..................................................3
   3.1 Waiting for the minimum amount of digits.....................3
   3.2 Sending the first INVITE.....................................3
   3.3 Sending overlap signalling to the SIP network................4
   3.4 Applicability of this mechanism..............................5
   3.5 Receiving multiple responses.................................5
   3.6 Canceling pending INVITE transactions........................6
   3.7 INVITEs reaching multiple gateways...........................6
   4   SIP to ISUP..................................................6
   4.1 Receiving subsequent INVITEs.................................6
   5   Conclusions..................................................6
   6   Acknoledgements..............................................7
   7   References...................................................7
   8   Authors³ addresses...........................................7

1. Introduction

   A mapping between the Session Initiation Protocol (SIP) [1] and the
   ISDN User Part (ISUP) [2] of SS7 is described in [3]. However, [3]
   just takes into consideration ISUP en-bloc signalling. En-bloc
   signalling consists of sending the complete telephone number of the
   callee in the first signalling message. Although modern switches
   always use en-bloc signalling, some parts of the PSTN still use
   overlap signalling. Overlap signalling consists of sending just some
   digits of the callee³s number in the first signalling message.
   Further digits are sent in subsequent signalling messages.

2. Overlap signalling in SIP

   SIP uses en-bloc signalling. The Request-URI of an INVITE message
   contains the whole address of the callee. Even if the Request-URI
   contains a tel URI instead of a SIP URI, the INVITE contains the
   whole number. Breaking this principle would just bring undesirable
   problems to network designers. Therefore, it is strongly recommended
   not to use any kind of overlap signalling in a SIP network. The
   recommended behavior is to convert overlap signalling to en-bloc at
   the edge of the network and then use en-bloc signalling in SIP. A
   gateway connected to a part of the PSTN where overlap signalling is
   used can perform this conversion through the use of timers.

   However, although its use is discouraged, some applications need to
   use overlap signalling in order to meet service requirements (i.e.
   establishment time). Such applications should use the mechanism
   described in this document. This document also describes in which
   scenarios is acceptable to use such a mechanism and when, on the
   other hand, it is completely unacceptable to use overlap.

Camarillo/Roach/Peterson/Ong                                         2

               Mapping of ISUP Overlap Signalling to SIP

3. ISUP to SIP

   In this scenario the gateway receives an IAM (Initial Address
   Message) that contains just a portion of the called number. The rest
   of the digits dialed arrive later in one or more SAMs (Subsequent
   Address Message).

3.1 Waiting for the minimum amount of digits

   If the IAM contain less than the minimum amount of digits to route a
   call, the gateway starts T35 and waits until the minimum amount of
   digits that can represent a telephone number is received (or a stop
   digit is received). If T35 expires before the minimum amount of
   digits (or a stop digit) has been received a REL with cause value 28
   is sent to the ISUP side.

   If a stop digit is received the INVITE message generated by the
   gateway will contain the complete called number. Therefore, the call
   proceeds as usual - no overlap signalling in the SIP network.

3.2 Sending the first INVITE

   There are cases when the gateway, after having received the
   minimum amount of digits, cannot know whether the number received is
   a complete number or not. Since supporting overlap signalling in the
   SIP network is an option that may be deemed undesirable, the gateway
   may elect to collect digits until a timer (T10) expires or a stop
   digit (such as #) is entered by the user  (note that T10 is
   refreshed every time a new digit is received).

   In this case, when T10 expires, an INVITE with the digits collected
   so far is sent to the SIP side. After this, any SAM received is

        PSTN                      MGC/MG                       SIP
          |                          |                          |
          |-----------IAM----------->| Starts T10               |
          |                          |                          |
          |-----------SAM----------->| Starts T10               |
          |                          |                          |
          |-----------SAM----------->| Starts T10               |
          |                          |                          |
          |                          |                          |
          |             T10 expires  |---------INVITE---------->|
          |                          |                          |

   Note that T10 is defined for conversion between CAS signalling and
   en-bloc ISUP. PSTN switches usually implement an equivalent
   proprietary timer to convert overlap ISUP to en-bloc ISUP. This

Camarillo/Roach/Peterson/Ong                                         3

               Mapping of ISUP Overlap Signalling to SIP

   document uses T10 and does not define a new timer because T10 seems
   suitable for overlap to SIP conversion.

3.3 Sending overlap signalling to the SIP network

   Although the behavior just described is recommended by this
   document, a gateway might still decide to send overlap signalling in
   the SIP network. In this case, the gateway should proceed as

   As soon as the minimum amount of digits is received an INVITE is
   sent and T10 is started. This INVITE is built following the
   procedures described in [3].

   If a SAM arrives T10 is refreshed and a new INVITE with the new
   digits received is sent. The new INVITE has the same Call-ID and the
   same From header field including the tag as the first INVITE sent,
   but has an updated Request-URI. The new Request-URI contains all the
   digits received so far. The To header field of the new INVITE
   contains all the digits as well, but has no tag.

        Note that it is possible to receive a response to the first
        INVITE before having sent the second INVITE. In this case, the
        response received would contain a To tag and information
        (Record-Route and Contact) to build a Route header field. The
        new INVITE to be sent (containing new digits) should not use
        any of these headers. That is, the new INVITE does not contain
        neither To tag nor Route header field. This way this new INVITE
        can be routed dynamically by the network providing services
        (see Section 3.7).

   The new INVITE should, of course, contain a Cseq field. It is
   recommended that the Cseq of the new INVITE is higher than any of
   the previous Cseq that the gateway has generated for this Call-ID
   (no matter for which dialog the Cseq was generated).

        When an INVITE forks responses from different locations might
        arrive establishing one or more early dialogs. New requests
        such as PRACK or COMET can be sent within every particular
        early dialog. This implies that the Cseq number spaces of
        different early dialogs are different. Sending a new INVITE
        with a Cseq that is still unused by any of the remote
        destinations avoids confusion at the destination.

   If the gateway is encapsulating ISUP messages as SIP bodies, it
   should place the IAM and all the SAMs received so far in this

         PSTN                      MGC/MG                       SIP
          |                          |                          |
          |-----------IAM----------->| Starts T10               |
          |                          |---------INVITE---------->|
          |                          |                          |

Camarillo/Roach/Peterson/Ong                                         4

               Mapping of ISUP Overlap Signalling to SIP

          |-----------SAM----------->| Starts T10               |
          |                          |---------INVITE---------->|
          |                          |                          |
          |-----------SAM----------->| Starts T10               |
          |                          |---------INVITE---------->|
          |                          |                          |

   If 4xx, 5xx or 6xx final responses arrive (e.g. 484 address
   incomplete) for the pending INVITE transactions before T10 has
   expired the gateway should not send any REL. A REL is sent just if
   no more SAMs arrive, T10 expires and all the INVITEs sent have been
   answered with a final response (different than 200 OK).

         PSTN                      MGC/MG                       SIP
          |                          |                          |
          |-----------IAM----------->| Starts T10               |
          |                          |---------INVITE---------->|
          |                          |<---------484-------------|
          |                          |----------ACK------------>|
          |                          |                          |
          |                          |                          |
          |             T10 expires  |                          |
          |<----------REL------------|                          |

   The best status code among all the responses received for all the
   INVITEs that were generated is used to calculate the cause value of
   the REL as described in [3].

        The computation of the best response is done in the same way as
        forking proxies compute the best response to be returned to the
        client for a particular INVITE. Note that the best response is
        not always the response to the INVITE that contained more
        digits. If the user dials a particular number and then types an
        extra digit by mistake a 486 (Busy Here) could be received for
        the first INVITE and a 484 (Address Incomplete) for the second
        one (which contained more digits).

3.4 Applicability of this mechanism

   This mechanism is applicable only under certain circumstances. A
   ingress gateway may use overlap signalling in SIP only if an
   analysis of the called party number shows that it belongs to a part
   of the PSTN where overlap signalling is used. This ensures that a
   particular prefix of the number does not identify any other user.

   Within some dialing plans in the PSTN, a phone number might be a
   prefix of another one. This situation is not common, but it can
   certainly occur. Where en-bloc signalling is used, this ambiguity is
   resolved before the digits are placed in the en-bloc signalling. If
   overlap signaling was used in this situation, a different user than
   the one the caller intended to call might be contacted.

Camarillo/Roach/Peterson/Ong                                         5

               Mapping of ISUP Overlap Signalling to SIP

3.5 Receiving multiple responses

   When overlap signalling in SIP is used the ingress gateway sends
   multiple INVITEs. Accordingly, it will receive multiple responses.
   The responses to all the INVITEs sent except for one (normally, but
   not necessarily the last one) are typically 400 class responses
   (e.g. 484 Address Incomplete or 490 Request Updated) that terminate
   the INVITE transaction.

   However, a 183 Session Progress response with a media description
   can also be received. The media stream will typically contain a
   message such as "The number you have just dialed does not exist".

   The issue of receiving different 183 Session Progress responses with
   media descriptions does not only apply to overlap signalling. When
   vanilla SIP is used, several responses can also arrive to a gateway
   if the INVITE forked. It is then up to the gateway to decide which
   media stream should be played to the user.

   However, overlap signalling adds a requirement to this process. As a
   general rule, a media stream corresponding to the response to an
   INVITE with a greater number of digits should be given more priority
   than media streams from responses with less digits.

3.6 Canceling pending INVITE transactions

   When a gateway sends a new INVITE containing new digits, it should
   not CANCEL the previous INVITE transaction. This CANCEL could arrive
   before the new INVITE to an egress gateway and trigger a REL before
   the new INVITE arrived. INVITE transactions are typically terminated
   by the reception of 4xx responses.

   However, once a 200 OK response has been received, the gateway
   should CANCEL all the other INVITE transactions were generated. A
   particular gateway might implement a timer to wait for some time
   before sending any CANCEL. This gives time to all the previous
   INVITE transactions to terminate smoothly without generating more
   signalling traffic (CANCEL messages).

3.7 INVITEs reaching multiple gateways

   Since every new INVITE sent by a gateway represents a new
   transaction they can be routed in different ways. For instance, the
   first INVITE might be routed to a particular gateway and a
   subsequent INVITE to another. The result is that both gateways
   generate an IAM. Since one of the IAMs (or both) has an incomplete
   number, it would fail, having already consumed PSTN resources.

   It has been proposed to make all the INVITEs follow the same path as
   the first one. This proposal would resolve the problem of having
   INVITEs hitting different gateways, but would restrict the number of
   services the SIP network can provide. It would not be possible to

Camarillo/Roach/Peterson/Ong                                         6

               Mapping of ISUP Overlap Signalling to SIP

   route a subsequent INVITE to an application server just because the
   previous one was routed in a different way.

   This issue should be taken into consideration before using overlap
   signalling in SIP. If sending multiple IAMs to the PSTN is not
   acceptable in a particular domain, overlap signalling should not be

4. SIP to ISUP

   In this scenario the gateway receives multiple INVITEs that have the
   same Call-ID but have different Request-URIs.

        Note that these INVITEs do not belong to the same dialog
        because they have different To header fields.

4.1 Receiving subsequent INVITEs

   An egress gateway does not have any means to know whether SIP
   overlap signalling is being used or not. So, upon reception of an
   INVITE, the gateway generates an IAM following the procedures
   described in [3].

   If a gateway receives a subsequent INVITE with the same Call-ID and
   From tag as the previous one and an updated Request-URI, a SAM
   should be generated as opposed to a new IAM. Upon reception of a
   subsequent INVITE, the INVITE received previously is answered with
   490 Request Updated.

   If the gateway is attached to the PSTN in an area where en-bloc
   signalling is used, a REL for the previous IAM and a new IAM should
   be generated.

5. Conclusions

   The mechanism described in this document is intended to be used in a
   close environment. Using it in an open network such as the Internet
   would cause problems such as multiple IAMs generated. If this
   mechanism was used with telephone numbers that belong to an en-bloc
   zone, calls could end up reaching a different callee than the one
   who was supposed to receive the call.

   Due to these problems, it is strongly recommended that this
   mechanism is only used if a particular application must fulfil
   strong requirements regarding establishment delay. Otherwise, the
   ingress gateway should always perform overlap to en-bloc conversion.

6. Acknowledgments

   The authors would like to thank Jonathan Rosenberg, Olli Hynonen and
   Mike Pierce for their feedback on this document.

7. References

Camarillo/Roach/Peterson/Ong                                         7

               Mapping of ISUP Overlap Signalling to SIP

   [1] M. Handley, H. Schulzrinne, E. Schooler, J. Rosenberg, "SIP:
   Session Initiation Protocol", RFC 2543, IETF; March 1999.

   [2] "Application of the ISDN user part of CCITT signaling system No.
   7 for international ISDN interconnections" ITU-T Q.767
   recommendation, February 1991.

   [3] G. Camarillo, A. Roach, J. Peterson, L. Ong, "ISUP to SIP
   Mapping", draft-ietf-sip-isup-01.txt, IETF; May 2001. Work in

8. Authors³ Addresses

   Gonzalo Camarillo
   Advanced Signalling Research Lab
   FIN-02420 Jorvas
   Phone: +358 9 299 3371
   Fax: +358 9 299 3052

   Adam Roach
   Ericsson Inc.
   Mailstop L-04
   851 International Pkwy.
   Richardson, TX 75081
   Phone: +1 972-583-7594
   Fax: +1 972-669-0154

   Jon Peterson
   NeuStar, Inc
   1800 Sutter St Suite 570
   Concord, CA 94520

   Lyndon Ong
   10480 Ridgeview Court
   Cupertino, CA 95014

   Full Copyright Statement

   Copyright (c) The Internet Society (2001). All Rights Reserved.

Camarillo/Roach/Peterson/Ong                                         8

               Mapping of ISUP Overlap Signalling to SIP

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph
   are included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an

Camarillo/Roach/Peterson/Ong                                         9