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Internet Engineering Task Force                     Gonzalo Camarillo
Internet draft                                             Adam Roach
<draft-ietf-sip-overlap-01.txt>                              Ericsson
August 2001
Expires: February 2002                                   Jon Peterson
                                                              NeuStar

                                                           Lyndon Ong
                                                                Ciena


               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
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Abstract

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


















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               Mapping of ISUP Overlap Signalling to SIP


TABLE OF CONTENTS

   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.



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               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 and an INVITE with the digits
   collected so far is sent to the SIP side. After this, any SAM
   received is ignored.


        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


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               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
   follows.

   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 tag as the first INVITE sent, but has an updated Request-
   URI field. The new INVITE contains no To 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. 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. 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 call-leg the Cseq was generated).

        When an INVITE forks responses from different locations might
        be received. New requests such as PRACK, COMET or INVITEs
        negotiating early media can be sent to every particular remote
        location. This implies that the Cseq number spaces of different
        call-legs within the same call 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
   INVITE.

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

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               Mapping of ISUP Overlap Signalling to SIP


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


   If class 4, 5 or 6 final responses arrives (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 status code of the response to the last INVITE sent by the
   gateway (the one that contained more digits) is used to calculate
   the cause value of the REL as described in [3].

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.

   When en-bloc signalling is used in the PSTN a phone number might be
   a prefix of another one. This situation is not common, but it can
   certainly occur. If overlap signalling was used in this situation a
   different user than the one the caller intended to call might be
   contacted.

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 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".


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               Mapping of ISUP Overlap Signalling to SIP


   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. 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 400 class responses.

   However, once a 200 OK response has been received, the gateway
   should CANCEL all the previous INVITE transactions that did not
   contain enough digits. 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
   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
   used.

4. SIP to ISUP

   In this scenario the gateway receives multiple INVITEs that belong
   to the same call (same Call-ID) but have different Request-URIs.

4.1 Receiving subsequent INVITEs

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               Mapping of ISUP Overlap Signalling to SIP



   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 and Olli Hynonen
   for their feedback on this document.

7. References

   [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
   progress.

8. Authors³ Addresses

   Gonzalo Camarillo
   Ericsson
   Advanced Signalling Research Lab

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               Mapping of ISUP Overlap Signalling to SIP


   FIN-02420 Jorvas
   Finland
   Phone: +358 9 299 3371
   Fax: +358 9 299 3052
   Email: Gonzalo.Camarillo@ericsson.com

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

   Jon Peterson
   NeuStar, Inc
   1800 Sutter St Suite 570
   Concord, CA 94520
   USA
   E-Mail: jon.peterson@neustar.com

   Lyndon Ong
   Ciena
   10480 Ridgeview Court
   Cupertino, CA 95014
   E-Mail: lyOng@ciena.com




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               Mapping of ISUP Overlap Signalling to SIP


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