Internet Draft James Yu
Document: <draft-yu-tel-url-03.txt> NeuStar, Inc.
Category: Standard Track November 2001
Extensions to the "tel" and "fax" URLs to Support
Number Portability and Freephone Service
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026[1].
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ABSTRACT
This document proposes some extensions to the "tel" and "fax" Uniform
Resource Locators (URLs) for supporting number portability (NP) and
freephone service. Those proposed extensions allow the Session
Initiation Protocol (SIP) to carry those URLs or to convert those
URLs to the SIP URL so as to support NP and freephone service. The
proposed extensions 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 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 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 some extensions to the "tel" and "fax"
Uniform Resource Locators (URLs)[4] for supporting NP and freephone
service allowing the Session Initiation Protocol (SIP) to carry
those URLs or to convert those URLs to the SIP URL. The proposed
extensions 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 associated 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 extensions to the "tel"
and "fax" URLs to support NP and freephone service, and those
proposed extensions are detailed in sections 5 and 6. Section 7
gives a few examples as to how those proposed extensions are used.
Section 8 discusses the signaling interworking. Section 9 lists the
major changes from the previous version of this document followed by
the conclusion.
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2. Abbreviations
ABNF Augmented Backus-Naur Form
ANSI American National Standards Institute
CIC Carrier Identification Code (also cic)
CIP Carrier Identification parameter
FCI Forward Call Indicator
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
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, an indicator that indicates whether a query to the
NP Database (NPDB) has been performed, and a location number that
identifies the location of the originating switch.
The dialed called party 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 called party number. The NPDB dip
indicator informs the network entities downstream towards the
terminating network (e.g., the network that currently serves the
called party 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" or
"fax" URL before the NPDB dip is performed, it stays at the same
place (i.e., right after the "tel:" or "fax:"). 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
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used to carry any routing number information that is different from
the directory number (e.g., carried right after the "tel:") even
when NP is not involved.
The second new parameter "npdi," which stands for "NPDB dip
indicator," indicates whether NPDB dip has been performed.
These two new parameters are added to the "tel" and "fax" URLs
following the rules defined for "future-extension" for the "global-
phone-number" and "local-phone-number."
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:" or
"fax:" 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" and "fax" URLs following the rules
defined for "future-extension" for the "global-phone-number" and
"local-phone-number."
5. Proposed Extensions to the "tel" URL Scheme
The proposed extensions are to be added to "global-phone-number" and
"local-phone-number" based on Augmented Backus-Naur Form (ABNF)[5].
Only the impacted items and new items are shown below.
global-phone-number = "+" base-phone-number [isdn-subaddress]
[post-dial]
*1(";" routing-number) ;new ext.
*1(";" npdb-dip-indicator) ;new ext.
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*1(";" carrier-id-code) ;new ext.
*(area-specifier / service-provider /
future-extension)
local-phone-number = 1*(phonedigit / dtmf-digit /
pause-character) [isdn-subaddress]
[post-dial] area-specifier
*1(";" routing-number) ;new ext.
*1(";" npdb-dip-indicator) ;new ext.
*1(";" carrier-id-code) ;new ext.
*(area-specifier / service-provider /
future-extension)
routing-number = rn-tag "=" *1("+") rn-ident
rn-tag = "rn"
rn-ident = *(hex excluding "F" / visual-separator)
npdi-dip-indicator = npdi-tag "=" npdi-ident
npdi-tag = "npdi"
npdi-ident = "yes" / "no"
carrier-id-code = cic-tag "=" *1("+") cic-ident
cic-tag = "cic"
cic-ident = *phonedigit
It is assumed that national routing number may appear with other
global-phone-number information and international routing number may
appear with other local-phone-number information. The routing
number digit can be any hexadecimal digit except the digit "F."
The first 1-3 digits in the "cic" 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" and/or "rn" 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 (equivalent to npdi=no) 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 "yes" in the
"npdi" parameter, it will perform the NPDB query. The NPDB query is
outside the scope of this document. The routing number received in
the response (plus the "+" and the country code if a national number
is received in the response) 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 set
to "yes" 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
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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 extensions 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)[6] has
been defined. 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 "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 "CIC" in the U.S. currently identifies the inter-exchange
carrier that supports the POTS and/or freephone service. It can be
expanded to include VoIP carriers and local exchange 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 [7].
Please see section 8 for the discussion on the signaling
interworking between the GSTN ISUP and SIP (e.g., "sip" or "tel"
URL).
6. Proposed Extension to the "fax" URL Scheme
The proposed extensions are to be added to "global-phone-number" and
"local-phone-number" based on ABNF. Only the impacted items and new
items are shown below.
fax-global-phone = "+" base-phone-number [isdn-subaddress]
[t33-subaddress] [post-dial]
*1(";" routing-number) ;new ext.
*1(";" npdb-dip-indicator) ;new ext.
*1(";" carrier-id-code) ;new ext.
*(area-specifier / service-provider /
future-extension)
fax-local-phone = 1*(phonedigit / dtmf-digit /
pause-character) [isdn-subaddress]
[t33-subaddress] [post-dial]
area-specifier
*1(";" routing-number) ;new ext.
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*1(";" npdb-dip-indicator) ;new ext.
*1(";" carrier-id-code) ;new ext.
*(area-specifier / service-provider /
future-extension)
routing-number = rn-tag "=" *1("+") rn-ident
rn-tag = "rn"
rn-ident = *(hex excluding "F" / visual-separator)
npdi-dip-indicator = npdi-tag "=" npdi-ident
npdi-tag = "npdi"
npdi-ident = "yes" / "no"
carrier-id-code = cic-tag "=" *1("+") cic-ident
cic-tag = "cic"
cic-ident = *phonedigit
The same discussions in Section 5 also apply to this section.
7. Examples
7.1 NP Examples
To simply 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=yes 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).
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=yes SIP/2.0
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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.).
7.2 Freephone Service Examples
To simply 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.
7.3 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," for example, the "tel" URI 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 extensions to the "tel" URL to carry the NP- and freephone-
related information. Since the "fax" URL may be used for fax calls,
both the "tel" and "fax" URLs need to be enhanced to support NP and
freephone service. Some enhancements to the SIP protocol may be
required to fully support the NP and freephone service (e.g., to
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.
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Example 1: A "tel" URL such as
tel:+1-202-533-1234;rn=+1-202-544-0000;npdi=yes
can be converted to a "sip" URL shown below.
sip:+1-202-533-1234;rn=+1-202-544-0000;
npdi=yes@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
8. 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 are mapped/carried in the proper ISUP parameters. Some possible
mapping are briefly described here; however, the exact mapping
between the SIP and ISUP are defined by the "SIP for Telephony"
(SIP-T)[8,9], 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 "npdi" parameter that contains "yes" 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
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
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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 set "npdi" to "yes," 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=yes |
| Forward Call Indicator | |
+----------------------------------+----------------------+
| Carrier Identification Parameter | cic |
+----------------------------------+----------------------+
9. Security Considerations
This document does not introduce new security implications other
than those described in Section 5 of [4].
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10. IANA Considerations
The three extensions proposed in this document should be registered
with IANA as the extensions to tel URL [4].
11. Conclusion
This Internet Draft proposes some extensions to the "tel" and "fax"
URLs described in [4] to allow the SIP protocol to carry the NP- and
freephone service-related information in the "tel" and "fax" URLs.
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 URL. The extensions 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 extensions,
people surely will come up innovative ways of using SIP to support
many of the existing and new services. If those proposed extensions
are agreed, it is proposed to follow the standardization process to
issue this document as a RFC.
NORMATIVE REFERENCES
[1] S. Bradner, "The Internet Standards Process -- Revision 3,"
RFC2026, October 1996.
[3] J. Rosenberg, et al., "SIP: Session Initiation Protocol," draft-
ietf-sip-rfc2543bis-05.txt, October 2001.
[4] A. Vaha-Sipila, "URLs for Telephone Calls," RFC 2806, April
2000.
[5] D. Crocker and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF," RFC 2234, November 1997.
[6] J. Rosenberg, H. Salama and M. Squire, RFC XXX, "Telephony
Routing Information Protocol (TRIP)," November 2001.
[7] 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.
INFORMATIVE REFERENCES
[2] M. Foster, T. McGarry and J. Yu, "Number Portability in the
GSTN: An Overview," draft-ietf-enum-e164-gstn-np-03.txt,
November 2001.
[8] A. Vemuri and J. Peterson, draft-vemuri-sip-t-context-02.txt,
"SIP for Telephones (SIP-T): Context and Architectures,"
February 2001.
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[9] G. Camarillo, et al., draft-ietf-sip-isup-03.txt, "ISUP to SIP
Mapping," August 2001.
ACKNOWLEDGEMENT
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 sip/tel URL.
Authors' Address
James Yu
NeuStar, Inc.
1120 Vermont Avenue, NW, Suite 400
Washington, D.C., 20005
U.S.A.
Phone: +1-202-533-2814
Email: james.yu@neustar.com
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