One document matched: draft-yu-tel-url-01.txt
Differences from draft-yu-tel-url-00.txt
Internet Draft James Yu
Document: <draft-yu-tel-url-01.txt> NeuStar, Inc.
Category: Informational November 16, 2000
Extensions to the "tel" and "fax" URLs to Support
Number Portability and Freephone Service
<draft-yu-tel-url-01.txt>
Status of this Memo
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all provisions of Section 10 of RFC2026[1].
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NP and Freephone Service
ABTRACT
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.
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 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 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 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 extensions to the "tel"
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NP and Freephone Service
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.
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
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
oln Originating Location Number
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
tfn Translated-From-Number
TNS Transit Network Selection
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
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NP and Freephone Service
called party number) that NPDB dip has been performed; therefore,
there is no need to dip the NPDB again. The "originating location
number" is needed because the calling party number may not reveal
the originating switch's location in the NP environment (e.g.,
wireless roaming and location portability). The "originating
location number" is used for identifying the originating switch.
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:"). Three new
parameters are then required to support NP.
One new parameter is "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 the directory number (e.g., carried right after the
"tel:") even when NP is not involved.
The second new parameter is "npdi," which stands for "NPDB dip
indicator," indicates whether NPDB dip has been performed.
The third parameter "oln" standing for "originating location number"
identifies the location of the originating switch.
These three 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, the translated POTS number and
the "originating location 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 "originating location number" may be sent in
the SIP INVITE message to a redirect server or an application server
that performs number translation/call redirection service where the
call routing may be based on the location of the call origination.
Please note that the existing freephone service in the U.S. may use
a parameter such as the Calling Party Number or the Charge Number
for determining the location of the call origination.
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.
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A new parameter "tfn," which stands for "translated-from-number,"
carries the original dialed freephone number. This parameter, when
needed, can also be used to carry the directory number that is used
for number translation by any number translation service other than
the freephone service.
Another 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). There is
another ISUP parameter called Transit Network Selection (TNS) that
is used to identify the international carrier for handling an
international call. Whether the "cic" can be used or another new
"tel" extension is need to carry the TNS information is for further
study.
The "oln" parameter mentioned in Section 3 may be used to support
the freephone service. So there is no need to define a new
parameter for the freephone service.
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."
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.
*1(";" orig-location-number) ;new ext.
*1(";" carrier-id-code) ;new ext.
*1(";" translated-from-number) ;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(";" orig-location-number) ;new ext.
*1(";" carrier-id-code) ;new ext.
*1(";" translated-from-number) ;new ext.
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*(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"
orig-location-number = oln-tag "=" *1("+") oln-ident
oln-tag = "oln"
oln-ident = *DIGIT
carrier-id-code = cic-tag "=" *1("+") cic-ident
cic-tag = "cic"
cic-ident = *DIGIT
translated-from-number = tfn-tag "=" *1("+") tfn-ident
tfn-tag = "tfn"
tfn-ident = *DIGIT
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 "rn," "npdi," "oln," "cic" and "tfn" can appear at most once if
present. The "cic" and "oln" may be removed when there is no need
to carry them further in the call signaling messages. For example,
when a freephone call reaches the freephone service provider/carrier
serving that freephone number, the "cic" and "oln" may no longer be
needed when the call is 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
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.
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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] needs
to be 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.
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(";" orig-location-number) ;new ext.
*1(";" carrier-id-code) ;new ext.
*1(";" translated-from-number) ;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.
*1(";" npdb-dip-indicator) ;new ext.
*1(";" orig-location-number) ;new ext.
*1(";" carrier-id-code) ;new ext.
*1(";" translated-from-number) ;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"
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npdi-ident = "yes" / "no"
orig-location-number = oln-tag "=" *1("+") oln-ident
oln-tag = "oln"
oln-ident = *DIGIT
carrier-id-code = cic-tag "=" *1("+") cic-ident
cic-tag = "cic"
cic-ident = *DIGIT
translated-from-number = tfn-tag "=" *1("+") tfn-ident
tfn-tag = "tfn"
tfn-ident = *DIGIT
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 Request-Line of a complete 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, and the caller is served by a
switch identified by +1-703-456.
INVITE tel:+1-202-533-1234;oln=+1-703-456 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;oln=+1-703-456;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 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;oln=+1-703-456;npdi=yes SIP/2.0
Please note that it is legal to include the "rn" for carrying the
same called party number in the example described above; however, it
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is recommended not to include it to allow a simpler ISUP conversion,
if encountered.
7.2 Freephone Service Examples
To simply the examples and focus on the "tel" URL, only the Request-
Line of a complete SIP INVITE message is shown. A SIP proxy server
receives a call to a freephone number +1-800-123-4567 with an "oln"
equal to =+1-703-538. After an interrogation with the freephone
database, a CIC with a value of =+1-6789 is received. 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;oln=+1-703-538;cic=+1-6789 SIP/2.0
If the freephone number is mapped to a POTS number +1-202-256-1234,
the INVITE message would look like
INVITE tel:+1-202-256-1234;oln=+1-703-538;tfn=+1-800-123-4567
SIP/2.0
Please note that the translated POTS number is placed right after
"tel:" after the number translation, and the dialed freephone number
is placed in the "tfn" parameter. Although the "To" header may
contain the freephone number, it is in the author's opinion that it
is better to use the new "tfn" parameter to carry the directory
number used for number translation in case that multiple "tel"
and/or "sip" URLs may be involved (e.g., in the Contact header).
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, it is decided to use 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. 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;oln=+1-703-456;rn=+1-202-533-1234;npdi=yes
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can be converted to a "sip" URL shown below.
sip:+1-202-533-1234;oln=+1-703-456;rn=+1-202-533-1234;
npdi=yes@sip.abc.com
Example 2: A "tel" URL such as
tel:+1-800-123-4567;oln=+1-703-538;cic=+1-6789
can be converted to a "sip" URL shown below.
sip:+1-800-123-4567;oln=+1-703-538;cic=+1-6789@sip.xyz.com
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)[7,8], 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 if a special value has been assigned to indicate
concatenated numbers. If to be carried further the "cic" can be
mapped to the ISUP Carrier Identification Parameter (CIP), and the
"oln" can be mapped to the ISUP Jurisdiction Information Parameter
(JIP) in the U.S.
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
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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 is set to "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 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 Europe), 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 | |
+----------------------------------+----------------------+
| Jurisdiction Information | oln |
+----------------------------------+----------------------+
| Carrier Identification Parameter | cic |
+----------------------------------+----------------------+
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9. Major Changes from the Previous Version
The major changes from version draft-yu-tel-url-00.txt are:
- "originating location number" or "oln" is added to support NP.
- "rn" and "npdi" are separately defined.
- Support for freephone service is included.
- The upper limit of 15 for "rn" is removed.
- Specify the precedence in call routing when the "cic," "rn" and
POTS number are all in the same "tel" or "sip" URL.
- Abbreviations are added.
- More examples are given including the conversion from the "tel"
URL to the "sip" URL.
10. 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, the Contact header in the response such as
"302 Moved" can carry one or more URLs. The extensions proposed in
this document also apply to the "tel" or "sip" URL(s) 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.
REFERENCES
[1] Scott Bradner, RFC2026, "The Internet Standards Process --
Revision 3," October 1996.
[2] M. Foster, T. McGarry and J. Yu, "Number Portability in the
GSTN: An Overview," draft-foster-e164-gstn-np-01.txt, July 2000.
[3] M. Handley, H. Schulzrinne, E. Schooler and J. Rosenberg, "SIP:
Session Initiation Protocol," draft--ietf-sip-rfc2543bis-00.ps,
May 2000.
[4] A. Vaha-Sipila, "URLs for Telephone Calls," RFC 2806, April
2000.
<draft-yu-tel-url-01> Informational - Expiration in May 15, 2001 12
Extension to the "tel" and "fax" URLs to Support November 16, 2000
NP and Freephone Service
[5] D. Crocker and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF," RFC 2234, November 1997.
[6] J. Rosenberg, H. Salama and M. Squire, draft-ietf-iptel-trip-
02.txt, "Telephony Routing Information Protocol (TRIP)," May
2000.
[7] A. Vemuri and J. Peterson, draft-vemuri-sip-t-context-00.txt,
"SIP for Telephones (SIP-T): Context and Architectures," July
14, 2000.
[8] F. Camarillo and A. Roach, draft-camarillo-sip-isup-bcp-00.txt,
"Best Current Practice for ISUP to SIP Mapping," March 2000.
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 and ISUP
interworking.
Authors' Address
James Yu
NeuStar, Inc.
1120 Vermont Avenue, NW, Suite 550
Washington, D.C., 20005
U.S.A.
Phone: +1-202-533-2814
Email: james.yu@neustar.com
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<draft-yu-tel-url-01> Informational - Expiration in May 15, 2001 13
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