Internet Draft                                                 James Yu
Document: <draft-yu-tel-url-08.txt>                       NeuStar, Inc.
Category: Standards Track                             November 19, 2003


              New Parameters for the "tel" URL to Support
                Number Portability and Freephone Service

                      <draft-yu-tel-url-08.txt>


Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026[1].

   Internet-Drafts are working documents of the Internet Engineering
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Copyright Notice

   Copyright (C) The Internet Society (2003).  All rights reserved.


ABTRACT

  This document proposes three parameters to the "tel" Uniform Resource
  Locator for supporting number portability (NP) and freephone service.
  Those proposed parameters allow the Session Initiation Protocol to
  carry the tel URL or to convert the tel URL to the SIP URL so as to
  support NP and freephone service.  The proposed parameters allow the
  SIP protocol to be used to derive the routing number for the ported
  geographical numbers, identify the freephone service provider/carrier
  or the Plain Old Telephone Service (POTS) number for a freephone
  number, and carry the NP- and freephone-related information in the
  SIP messages.



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1. Introduction

   Number portability (NP)[2] allows the telephone subscribers to keep
   their telephone numbers when they change service provider, move to a
   new location, or change the subscribed services.  The NP
   implementations in many countries presently support service provider
   portability for geographic numbers and some non-geographical
   numbers.  It has been identified that NP has impacts on several
   works-in-progress at the IETF.  One of the impacts is the need to
   carry the NP related information in the Session Initiation Protocol
   (SIP)[3] INVITE message after the NP database dip has been
   performed.

   Freephone service allows the called party to pay for the call by
   using special numbering blocks (e.g., 800, 888 and 877 number blocks
   in the U.S.) and requiring a translation from the special numbers to
   the Plain Old Telephone Service (POTS) numbers.  For countries that
   support freephone number portability using centralized databases to
   manage the number porting, the originating network usually performs
   a database dip to identify the freephone service provider/carrier
   that serves a particular freephone number so that it can route the
   freephone call to that freephone service provider/carrier.  If the
   originating network is the freephone service provider for that
   freephone number or is authorized by the freephone service
   provider/carrier for that freephone number, it translates the
   freephone number to a POTS number or some proprietary routing
   information based on certain algorithms for call routing.

   This document proposes three parameters to the "tel" Uniform
   Resource Locator (URL)[4] for supporting NP and freephone service
   allowing the Session Initiation Protocol (SIP) to carry the tel URL
   or to convert the tel URL to the SIP URL.  The proposed parameters
   may allow the SIP to be used to derive the routing number for the
   ported geographical numbers, to identify the freephone service
   provider/carrier or the Plain Old Telephone Service (POTS) number
   associate with a freephone number, and to carry the NP and
   freephone-related information in the SIP messages.

   Section 2 below lists the abbreviations used in this document.
   Sections 3 and 4 describe the need for the parameters to the "tel"
   URL to support NP and freephone service correspondingly, and those
   proposed parameters are detailed in sections 5.  Section 6 gives a
   few examples as to how those proposed parameters are used.  Section
   7 discusses the signaling interworking between the IP-based network
   and the traditional telephony network.  Section 8 is the conclusion.


2. Abbreviations

   ABNF   Augmented Backus-Naur Form
   ANSI   American National Standards Institute
   CIC    Carrier Identification Code (also cic)
   CIP    Carrier Identification parameter

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   FCI    Forward Call Indicator
   FGB    Feature Group B
   FGD    Feature Group D
   GAP    Generic Address Parameter
   GSTN   Global Switched Telephone Network
   IC     Identification Code
   IETF   Internet Engineering Task Force
   IP     Internet Protocol
   ISUP   Integrated Services Digital Network User Part
   JIP    Jurisdiction Information Parameter
   LEC    Local Exchange Carrier
   NANPA  North American Numbering Plan Administration
   NP     Number Portability
   NPDB   Number Portability Database
   npdi   NPDB dip indicator
   PNTI   Ported Number Translation Indicator
   POTS   Plain Old Telephone Service
   rn     Routing Number
   SIP    Session Initiation Protocol
   SIP-T  SIP for Telephony
   SS7    Signaling System No. 7
   TRIP   Telephony Routing Information Protocol
   URI    Uniform Resource Identifier
   URL    Uniform Resource Locators


3. NP Support

   The NP-related information includes the dialed directory number, a
   routing number, and an indicator that indicates whether a query to
   the NP Database (NPDB) has been performed.

   The dialed directory number may be needed at the terminating switch
   so that the call can be terminated to the called party (e.g., a line
   card).  The routing number allows the network, either the Global
   Switched Telephone Network (GSTN) or the Internet Protocol (IP)-
   based network, to route the call to the network or switch that
   currently serves the dialed directory number. In some NP
   implementations, the routing number even identifies the line card
   that is associated with the dialed directory number.  The NPDB dip
   indicator informs the network entities downstream towards the
   terminating network (e.g., the network that currently serves the
   directory number) that NPDB dip has been performed; therefore, there
   is no need to dip the NPDB again.

   Since the dialed directory number is already present in the "tel"
   URL before the NPDB dip is performed, it stays at the same place
   (i.e., right after the "tel:").  Two new parameters are then
   required to support NP.

   The first parameter "rn," which stands for "routing number," carries
   the routing number used for call routing.  This parameter can be
   used to carry any routing number information that is different from

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   the directory number (e.g., carried right after the "tel:") even
   when NP is not involved.

   The second parameter "npdi," which stands for "NPDB dip indicator,"
   indicates whether NPDB dip has been performed.

   These two new parameters are added to the "tel" URL to support NP.


4. Freephone Service Support

   The freephone-related information includes the dialed freephone
   number, the carrier identification code (CIC) that identifies the
   freephone service provider/carrier and the translated POTS number.

   The dialed freephone number after number translation may need to be
   passed to the called party for purposes such as customer account
   management.  The CIC code is needed to identify the service
   provider/carrier that is to receive and process the freephone call.
   The translated POT number identifies the called party that is to
   receive the call.

   The translated POT number will be placed right after the "tel:" so
   there is no need for a new parameter to carry it.

   A new parameter "cic," which stands for carrier identification code,
   identifies the freephone service provider/carrier associated with
   the freephone number in question.  If a country uses the CIC codes
   to identify the service providers/carriers that are not limited to
   the freephone service providers/carriers, this new parameter can
   also be used to identify those service providers/carriers even when
   freephone service is not involved.   One example is the CIC dialed
   by the caller for selecting a specific inter-exchange carrier in the
   U.S. (e.g., 101XXXX).

   "cic" is added to the "tel" URL as the third parameter.


5. Proposed Parameters to the "tel" URL Scheme

   The following parameters are to be added to the tel URL based on
   Augmented Backus-Naur Form (ABNF)[5]:

                             *1(routing-number)
                             *1(npdb-dip-indicator)
                             *1(carrier-id-code)

   The proposed parameters are further described below.

   routing-number          = ";rn=" global-rn / local-rn
   global-rn               = "+" 1*phonedigit-hex
   local-rn                = 1*phonedigit-hex   [context]
   npdi-dip-indicator      = "npdi"

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   carrier-id-code         = "cic=" global-cic / local-cic
   global-cic              = "+" 1*phonedigit-hex
   local-cic               = 1*phonedigit-hex   [cic-context]
   cic-context             = ô;cic-context=ö descriptor

   The presence of ônpdiö indicates that NPDB dip has been performed.
   If ônpdiö is not present, it indicates that either NPDB dip is not
   yet performed or NP is not relevant.

   The first 1-3 digits in the ôglobal-cicö identify a country code.
   The rest of the digits identify a carrier ID code assigned in that
   country.

   The "rn," "npdi," and "cic" can appear at most once if present.  The
   "cic," "rn" or ônpdiö may be removed when there is no need to carry
   it further in the call signaling messages.  For example, when a
   freephone call reaches the freephone service provider/carrier
   serving that freephone number, the "cic" may no longer be needed
   when the call is to be routed to the called party or another
   network. Whether and when to remove the new parameters proposed in
   this document are outside the scope of this document.

   When the "rn" is present, the "npdi" may or may not be present.
   This is because that the routing number may be present independent
   of NP.  When the "npdi" parameter is not present, it indicates that
   either NPDB dip has not been performed or NP is not relevant.  If a
   SIP server is set to perform the NPDB queries and if a received
   INVITE message does not contain the "npdi" parameter, it will
   perform the NPDB query.  The NPDB query is outside the scope of this
   document.  Please see [6] for using SIP to access the NP data.  The
   routing number received in the response (converted to global- or
   local-rn format) will replace the routing number in the "rn"
   parameter if present or will be used by the new "rn" parameter if
   "rn" parameter is not present.  The "npdi" parameter will be
   included in this case.  The routing number can be a global routing
   number (e.g., with "+" and the country code plus the national
   number) or a local (e.g., network-specific) routing number.  It is
   also possible that the SIP protocol can be used for the NP query.
   In that case, the response (e.g., 302 Moved) to the SIP message may
   carry the NP related information in the "tel" or "sip" URL format
   with the parameters proposed in this document.

   Although it may be very rare but it is possible to have the "cic,"
   "rn" and POTS number all in the same "tel" URL.  When all the three
   are present, the "cic" is used for call routing.  A new address
   family in the Telephony Routing Information Protocol (TRIP)[7] has
   been defined for cic.  When only the "rn" and the POTS number are
   present, the "rn" is used for making routing decisions (e.g., check
   against the TRIP routing tables).  If the "cic" and "rn" parameters
   are not present, the telephone number right after "tel:" is used for
   call routing.  Please note that specific "cic" values can be
   reserved to indicate call routing information instead of a valid CIC
   that is assigned to a carrier.  For example, a "cic" value of "+1-

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   0110" in a response from the freephone database in the U.S.
   indicates "local, translated number provided."  In this particular
   case, the "cic" is ignored and the "rn" and the POTS number are used
   for call routing based on the rules described above.

   The "CICs" in the U.S. are assigned to entities that purchase
   Feature Group B (FGB) or Feature Group D (FGD) access, FGB
   translation access or are Local Exchange Carriers (LECs).  They are
   also returned in the response to a freephone query for identifying
   the freephone service provider that serves the queried freephone
   number.  The North American Numbering Plan Administration (NANPA)
   currently manages CIC assignment in the U.S.

   The "CIC" can be expanded to include VoIP carriers and other types
   of carriers in the same country or under the same country code so
   that all carriers can be identified in the IP domain for routing
   purpose.  International service providers and carriers can be
   identified by the E.164 country codes for global services and for
   Networks [8].

   Please see section 7 for the discussion on the signaling
   interworking between the GSTN ISUP and SIP (e.g., "sip" or "tel"
   URL).


6. Examples

6.1  NP Examples

   To simplify the examples and focus on the "tel" URL in the Request-
   URI, only the key information of the Request-Line in a SIP INVITE
   message is shown.  A SIP server receives an INVITE message as shown
   below where +1-202-533-1234 is the dialed called party number and
   has been ported out of the donor network.

        INVITE tel:+1-202-533-1234  SIP/2.0

   Assume that this SIP server is set to perform the NPDB query.  Since
   this INVITE message does not contain the "npdi" parameter, this SIP
   server will perform a NPDB query.  After receiving a successful
   response back from the queried NPDB, it formulates the following SIP
   INVITE message:

        INVITE tel:+1-202-533-1234;rn=+1-202-544-0000;
               npdi  SIP/2.0

   This SIP server then uses the "rn" parameter to make the routing
   decisions (e.g., using the routing number in the "rn" parameter to
   check against the TRIP tables to determine the terminating GSTN
   gateway).




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   The concept is that the "rn," if present, is used for making routing
   decisions, and the phone number after "tel:" is used for call
   routing only if the "rn" is not present.

   If the dialed called party number +1-202-533-1234 is not ported, the
   outbound SIP INVITE message may look like

        INVITE tel:+1-202-533-1234;npdi  SIP/2.0

   Please note that it may be legal to include the "rn" for carrying
   the called party number in the example described above; however, it
   is recommended not to include it because the called party number is
   not the same as the routing number (e.g., the Location Routing
   Number in the U.S.).


6.2  Freephone Service Examples

   To simplify the examples and focus on the "tel" URL, only the key
   information of the Request-Line in a SIP INVITE message is shown.  A
   SIP proxy server receives a call to a freephone number +1-800-123-
   4567.  After an interrogation with the freephone database, a CIC
   with a value of =+1-6789 is received ("+1" is added if not present
   in the response).  The CIC is used to route the freephone call
   further to the freephone service provider/carrier identified by the
   CIC.  Assume that the CIC code needs to be sent to the next SIP
   proxy server, the INVITE message would look like

        INVITE tel:+1-800-123-4567;cic=+1-6789 SIP/2.0

   If the freephone number is mapped to a POTS number +1-202-256-1234
   plus a cic of =+1-6789, the INVITE message would look like

        INVITE tel:+1-202-256-1234;cic=+1-6789  SIP/2.0

   Please note that the translated POTS number is placed right after
   "tel:" after the number translation.   Although the "To" header may
   contain the freephone number, there are cases where the freephone
   number (translated-from-number) may need to be passed in the tel URL
   or sip URL.  It is for further study.


6.3  Conversion from "tel" URL to "sip" URL

   The SIP INVITE message contains a "Request-URI" element that is used
   by the SIP servers for making routing decisions.  As indicated in
   [3], SIP servers may support Request-URIs with schemes other than
   "SIP" URL, for example, the "tel" URL scheme.  It is also known that
   anything that is defined for the "tel" URL can be converted to the
   SIP URL.  Therefore, the sip URL can automatically support the
   proposed parameters to the "tel" URL to carry the NP- and freephone-
   related information.  Some enhancements to the SIP protocol may be
   required to fully support the NP and freephone service (e.g., to

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   carry the "cic" information when the user portion does not carry a
   telephone number).  Those are outside the scope of this document.

   Two examples are shown below to show how a "tel" URL is converted to
   a "sip" URL.


   Example 1: A "tel" URL such as

        tel:+1-202-533-1234;rn=+1-202-544-0000;npdi

   can be converted to a "sip" URL shown below.

        sip:+1-202-533-1234;rn=+1-202-544-0000;
            npdi@sip.abc.com;user=phone

   Example 2: A "tel" URL such as

        tel:+1-800-123-4567;cic=+1-6789

   can be converted to a "sip" URL shown below.

        sip:+1-800-123-4567;cic=+1-6789@sip.xyz.com;user=phone


7. Interworking Between GSTN ISUP and SIP

   It is possible that interworking between SIP and Signaling System
   No. 7 (SS7) Integrated Services Digital Network User Part (ISUP) is
   required at the border between the GSTN and the IP-based network.
   For SIP to GSTN interworking and depending on the national ISUP
   support of NP and freephone service, the information in the "tel"
   URL is mapped/carried in the proper ISUP parameters.  Some possible
   mappings are briefly described here; however, the exact mapping
   between the SIP and ISUP are defined by the "SIP for Telephony"
   (SIP-T)[9,10], a mechanism that uses SIP to facilitate the
   interconnection of the GSTN with IP.  It is assumed that all the NP-
   and freephone-related parameters are present to simplify the
   discussion.  The interworking rules may be different if some
   parameters are not present.

   For the GSTN in the U.S., the routing number in the "rn" parameter
   is carried in the ISUP Called Party Number parameter.  The phone
   number after "tel:" is carried in the ISUP Generic Address Parameter
   (GAP) as the "ported number."  National numbers are usually carried
   (e.g., without the "+" and the country code) in the ISUP parameter.
   The presence of the "npdi" parameter causes the Ported Number
   Translation Indicator (PNTI) bit in the Forward Call Indicator (FCI)
   parameter to be set to "1."  If the terminating GSTN supports
   concatenated routing number and directory number (e.g., in Europe),
   then the routing number and the POTS number may be concatenated and
   put in the ISUP Called Party Number parameter.  The Nature of
   Address value will be set according to the terminating GSTN's

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   ISUP/NP standards (e.g., a special value is assigned to indicate
   concatenated numbers).  If to be carried further the "cic" can be
   mapped to the ISUP Carrier Identification Parameter (CIP).

   For GSTN to IP interworking, when the ISUP signaling contains the NP
   related information, the NP related information is mapped to the
   "tel" URL.  This happens for domestic calls where the originating
   GSTN has performed the NPDB query, or for international calls that
   have arrived at the terminating country's GSTN where that GSTN has
   performed the NPDB query.  It is assumed that the GSTN routes the
   call via the IP-based network to the terminating switch or network
   in the same country, and SIP and ISUP interworking is involved.  For
   the GSTN in the U.S., the interworking is straightforward.  The PNTI
   bit in the ISUP FCI parameter of "1" will cause "npdi" to be
   included," the number in the Called Party Number parameter plus the
   "+" and the country code, if a global routing number, is carried in
   the "rn" parameter, and the called party number in the Generic
   Address Parameter plus the "+" and the country code, if a global
   phone number, appears after "tel:".  For GSTN that supports
   concatenated routing number and directory number (e.g., in some
   European countries), the IP entity that performs the interworking
   may need to know the routing number used by the GSTN so that the
   routing number and the directory number in the concatenated format
   in the ISUP Called Party Number parameter can be separated and
   transported in the "rn" parameter and after "tel:" by adding the "+"
   and the country code to them if they are global routing number and
   phone number.  It is also possible to simply put the ISUP Called
   Party Number (with "+" and country code for a global phone number)
   after "tel:" without separating out the routing number and POTS
   number.

   The possible mapping between the American National Standards
   Institute (ANSI) ISUP and "tel" URL are summarized below.  It is
   assumed that all the information involved in the discussion is in
   the signaling message to simplify the discussion.  As indicated
   earlier, SIP-T is the one that defines the exact mapping.


      +----------------------------------+----------------------+
      |        ANSI ISUP                 |       "tel" URL      |
      +==================================+======================+
      |      Called Party Number         |          rn          |
      +----------------------------------+----------------------+
      |       "ported number"  in        |   POTS number after  |
      |    Generic Address Parameter     |         "tel:"       |
      +----------------------------------+----------------------+
      |    Ported Number Translation     |                      |
      |    Indicator bit set in the      |    npdi is present   |
      |     Forward Call Indicator       |                      |
      +----------------------------------+----------------------+
      | Carrier Identification Parameter |          cic         |
      +----------------------------------+----------------------+


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8. Conclusion

   This Internet Draft proposes three parameters to the "tel" URL
   described in [4] to allow the SIP protocol to carry the NP- and
   freephone service-related information in the "tel" URL.  There are
   several places in the SIP messages where URLs can be carried.  For
   example, each Contact header in the "302 Moved" response can carry
   one or more than one URL.  The parameters proposed in this document
   also apply to the "tel" or "sip" URL at those places in addition to
   the SIP Request-URI element.  With those parameters, people surely
   will come up innovative ways of using SIP to support many of the
   existing and new services.


9. Security Considerations

   In addition to those security implications discussed in the revised
   ôtelö URL [4], there are new security implications associated with
   the proposed parameters.

   If the value of the ôrnö or ôcicö is changed illegally when the SIP
   INVITE messages are en route to the destination entity, those
   messages may be routed to the wrong network or network element
   causing the sessions be rejected.

   If the ônpdiö is illegally inserted when the SIP INVITE messages are
   en route to the destination entity, those messages may be routed to
   the wrong network or network element causing the sessions be
   rejected.  It is less a problem if the ônpdiö is illegally removed.
   An additional NPDB query may be performed to retrieve the ôrnö
   information and have the ônpdiö included again.


10. IANA Considerations

   The three parameters proposed in this document are to be registered
   with IANA as the new parameters to the ôtelö URL [4].

   1. Parameter name û rn
      Applicability û used to carry a routing number (see Section 3)
      Mandatory or optional û optional
      Restrictions on syntax û see Section 5
      Reference to a specification û defined in this document

   2. Parameter name û npdi
      Applicability û its presence indicates that NPDB dip has been
      performed (see Section 3)
      Mandatory or optional û optional
      Restrictions on syntax û see Section 5
      Reference to a specification û defined in this document

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   3. Parameter name û cic
      Applicability û used to carry a Carrier ID Code (see Section 4)
      Mandatory or optional û optional
      Restrictions on syntax û see Section 5
      Reference to a specification û defined in this document

11. Normative References

   [1] Scott Bradner, RFC2026, "The Internet Standards Process --
       Revision 3," October 1996.

   [2] M. Foster, T. McGarry and J. Yu, RFC3482, "Number Portability in
       the GSTN: An Overview," February 2003.

   [3] J. Rosenberg, et al., RFC3261, "SIP: Session Initiation
       Protocol," June 2002.

   [5] D. Crocker and P. Overell, "Augmented BNF for Syntax
       Specifications: ABNF," RFC 2234, November 1997.

   [7] J. Rosenberg, H. Salama and M. Squire, RFC 3219, "Telephony
       Routing Information Protocol (TRIP)," January 2002.

   [8] ITU-T Rec. E.164.1, Criteria and procedures for the reservation,
       assignment, and reclamation of E.164 country codes and
       associated Identification Codes (ICs), March 1998.

   [9] A. Vemuri and J. Peterson, RFC3372, "SIP for Telephones (SIP-T):
       Context and Architectures," September 2002.

   [10] G. Camarillo, et al., RFC3398, " Integrated Services Digital
       Network (ISDN) User Part (ISUP) to Session Initiation Protocol
       (SIP) Mapping," December 2002.


12. Informative References

   [4] H. Schulzrinne and A. Vaha-Sipila, "The tel URI for Telephone
       Calls," draft-ietf-iptel-rfc2806bis-02.txt, June 29, 2003.

   [6] J. Yu, "Using SIP to Support NP and Freephone Service," draft-
       yu-sip-np-02.txt, January 3, 2003.


13. Acknowledgements

   The author would like to thank Penn Pfautz, Jon Peterson, Jonathan
   Rosenberg, Henning Schulzrinne and Antti Vaha-Sipila for the
   discussion of SIP support of NP and freephone service, ISUP
   interworking and/or sip/tel URL.



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14. Author's Address

   James Yu
   NeuStar, Inc.
   46000 Center Oak Plaza
   Sterling, VA 20166
   U.S.A.
   Phone: +1-571-434-5572
   Email: james.yu@neustar.biz



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<draft-yu-tel-url-08>         Expired on May 18, 2004         [Page 12]