One document matched: draft-yu-tel-url-08.txt
Differences from draft-yu-tel-url-07.txt
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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