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Versions: (draft-schulzrinne-ecrit-psap-callback) 00 01 02 03 04 05 06 07 09 10 11 12 13 RFC 7090

ECRIT                                                     H. Schulzrinne
Internet-Draft                                       Columbia University
Intended status: Standards Track                           H. Tschofenig
Expires: April 25, 2013                           Nokia Siemens Networks
                                                             C. Holmberg
                                                                Ericsson
                                                                M. Patel
                                             InterDigital Communications
                                                        October 22, 2012


             Public Safety Answering Point (PSAP) Callback
                 draft-ietf-ecrit-psap-callback-06.txt

Abstract

   After an emergency call is completed (either prematurely terminated
   by the emergency caller or normally by the call taker) it is possible
   that the call taker feels the need for further communication.  For
   example, the call may have been dropped by accident without the call
   taker having sufficient information about the current situation of a
   wounded person.  A call taker may trigger a callback towards the
   emergency caller using the contact information provided with the
   initial emergency call.  This callback could, under certain
   circumstances, be treated like any other call and as a consequence it
   may get blocked by authorization policies or may get forwarded to an
   answering machine.

   The IETF emergency services architecture specification already offers
   a solution approach for allowing PSAP callbacks to bypass
   authorization policies to reach the caller without unnecessary
   delays.  Unfortunately, the specified mechanism only supports limited
   scenarios.  This document discusses shortcomings of the current
   mechanisms and illustrates additional scenarios where better-than-
   normal call treatment behavior would be desirable.

   Note that this version of the document does not yet specify a
   solution due to the lack of the working group participants agreeing
   on the requirements.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-



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   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 25, 2013.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.




























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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  6
   3.  Callback Scenarios . . . . . . . . . . . . . . . . . . . . . .  7
     3.1.  Routing Asymmetry  . . . . . . . . . . . . . . . . . . . .  7
     3.2.  Multi-Stage Routing  . . . . . . . . . . . . . . . . . . .  8
     3.3.  Call Forwarding  . . . . . . . . . . . . . . . . . . . . .  9
     3.4.  Network-based Service URN Resolution . . . . . . . . . . . 11
     3.5.  PSTN Interworking  . . . . . . . . . . . . . . . . . . . . 12
   4.  SIP PSAP Callback Indicator  . . . . . . . . . . . . . . . . . 13
     4.1.  General  . . . . . . . . . . . . . . . . . . . . . . . . . 13
     4.2.  Usage  . . . . . . . . . . . . . . . . . . . . . . . . . . 13
     4.3.  Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . 13
       4.3.1.  General  . . . . . . . . . . . . . . . . . . . . . . . 13
       4.3.2.  ABNF . . . . . . . . . . . . . . . . . . . . . . . . . 13
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 14
     5.1.  Security Threat  . . . . . . . . . . . . . . . . . . . . . 14
     5.2.  Security Requirements  . . . . . . . . . . . . . . . . . . 14
     5.3.  Security Solution  . . . . . . . . . . . . . . . . . . . . 14
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 16
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 18
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 18


























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1.  Introduction

   Summoning police, the fire department or an ambulance in emergencies
   is one of the fundamental and most-valued functions of the telephone.
   As telephone functionality moves from circuit-switched telephony to
   Internet telephony, its users rightfully expect that this core
   functionality will continue to work at least as well as it has for
   the legacy technology.  New devices and services are being made
   available that could be used to make a request for help, which are
   not traditional telephones, and users are increasingly expecting them
   to be used to place emergency calls.

   An overview of the protocol interactions for emergency calling using
   the IETF emergency services architecture are described in [RFC6444]
   and [I-D.ietf-ecrit-phonebcp] specifies the technical details.  As
   part of the emergency call setup procedure two important identifiers
   are conveyed to the PSAP call taker's user agent, namely the Address-
   Of-Record (AoR), and, if available, the Globally Routable User Agent
   (UA) URIs (GRUU).  RFC 3261 [RFC3261] defines the AoR as:

      'An address-of-record (AOR) is a SIP or SIPS URI that points to a
      domain with a location service that can map the URI to another URI
      where the user might be available.  Typically, the location
      service is populated through registrations.  An AOR is frequently
      thought of as the "public address" of the user.'

   In SIP systems a single user can have a number of user agents
   (handsets, softphones, voicemail accounts, etc.) which are all
   referenced by the same AOR.  There are a number of cases in which it
   is desirable to have an identifier which addresses a single user
   agent rather than the group of user agents indicated by an AOR.  The
   GRUU is such a unique user- agent identifier, which is still globally
   routable.  RFC 5627 [RFC5627] specifies how to obtain and use GRUUs.
   [I-D.ietf-ecrit-phonebcp] also makes use of the GRUU for emergency
   calls.

   Regulatory requirements demand that the emergency call setup
   procedure itself provides enough information to allow the call taker
   to initiate a call back to the emergency caller.  This is desirable
   in those cases where the call got dropped prematurely or when further
   communication need arises.  The AoR and the GRUU serve this purpose.

   The communication attempt by the PSAP call taker back to the
   emergency caller is called 'PSAP callback'.

   A PSAP callback may, however, be blocked by user configured
   authorization policies or may be forwarded to an answering machine
   since SIP entities (SIP proxies as well as the SIP user equipment



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   itself) cannot differentiate the PSAP callback from any other SIP
   call.  "Call barring", "do not disturb", or "call diversion"(aka call
   forwarding) are features that prevent delivery of a call.  It is
   important to note that these features may be implemented by SIP
   intermediaries as well as by the user agent.

   Among the emergency services community there is the desire to offer
   PSAP callbacks a treatment such that chances are increased that it
   reaches the emergency caller.  At the same time a design must deal
   with the negative side-effects of allowing certain calls to bypass
   call forwarding or other authorization policies.  Ideally, the PSAP
   callback has to relate to an earlier emergency call that was made
   "not too long ago".  An exact time interval is difficult to define in
   a global IETF standard due to the variety of national regulatory
   requirements.

   To nevertheless meet the needs from the emergency services community
   a basic mechanism for preferential treatment of PSAP callbacks was
   defined in Section 13 of [RFC6444].  The specification says:

      'A UA may be able to determine a PSAP call back by examining the
      domain of incoming calls after placing an emergency call and
      comparing that to the domain of the answering PSAP from the
      emergency call.  Any call from the same domain and directed to the
      supplied Contact header or AoR after an emergency call should be
      accepted as a callback from the PSAP if it occurs within a
      reasonable time after an emergency call was placed.'

   This approach mimics a stateful packet filtering firewall and is
   indeed helpful in a number of cases.  It is also relatively simple to
   implement even though it requires state to be maintained by the user
   agent as well as by SIP intermediaries.  Unfortunately, the solution
   does not work in all deployment scenarios.  In Section 3 we describe
   cases where the currently standardized approach is insufficient.

















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2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

   Emergency services related terminology is borrowed from [RFC5012].
   This includes terminology like emergency caller, user equipment, and
   call taker.










































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3.  Callback Scenarios

   This section illustrates a number of scenarios where the currently
   specified solution, as specified in [I-D.ietf-ecrit-phonebcp], for
   preferential treatment of callbacks fails.  As explained in Section 1
   a SIP entity examines an incoming PSAP call back by comparing the
   domain of the PSAP with the destination domain of the emergency call.

3.1.  Routing Asymmetry

   In some deployment environments it is common to have incoming and
   outgoing SIP messaging routed through different SIP entities.
   Figure 1 shows this graphically whereby a VoIP provider uses
   different SIP proxies for inbound and for outbound call handling.
   Unless they two devices are state synchronized the callback hitting
   the inbound proxy would get treated like any other call since the
   emergency call established state information at the outbound proxy
   only.

































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                                                   ,-------.
                                                 ,'         `.
                      ,-------.                 /  Emergency  \
                    ,'         `.              |   Services    |
                   /  VoIP       \      I      |   Network     |
                  |   Provider    |     n      |               |
                  |               |     t      |               |
                  |               |     e      |               |
                  |   +-------+   |     r      |               |
               +--+---|Inbound|<--+-----m      |               |
               |  |   |Proxy  |   |     e      |   +------+    |
               |  |   +-------+   |     d      |   |PSAP  |    |
               |  |               |     i      |   +--+---+    |
     +----+    |  |               |     a-+    |      |        |
     | UA |<---+  |               |     t |    |      |        |
     |    |----+  |               |     e |    |      |        |
     +----+    |  |               |       |    |      |        |
               |  |               |     P  |   |      |        |
               |  |               |     r  |   |      |        |
               |  |   +--------+  |     o   |  |      |        |
               +--+-->|Outbound|--+---->v   |  |   +--+---+    |
                  |   |Proxy   |  |     i    | | +-+ESRP  |    |
                  |   +--------+  |     d    | | | +------+    |
                  |               |     e     || |             |
                  |               |     r     |+-+             |
                   \             /             |               |
                    `.         ,'               \             /
                      '-------'                  `.         ,'
                                                   '-------'

                  Figure 1: Example for Routing Asymmetry

3.2.  Multi-Stage Routing

   Consider the following emergency call routing scenario shown in
   Figure 2 where routing towards the PSAP occurs in several stages.  In
   this scenario we consider a SIP UA that uses LoST to learn the next
   hop destination closer to the PSAP.  This call is then sent to the
   user's VoIP provider.  The user's VoIP provider receives the
   emergency call and creates state based on the destination domain,
   namely state.com.  It then routes it to the indicated ESRP.  When the
   ESRP receives it it needs to decide what the next hop is to get it
   closer to the PSAP.  In our example the next hop is the PSAP with the
   URI psap@town.com.

   When a callback is sent from psap@town.com towards the emergency
   caller the call will get normal treatment by the VoIP providers
   inbound proxy since the domain of the PSAP does not match the stored



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   state information.



                                         ,-------.
       +----+                          ,'         `.
       | UA |--- esrp1@foobar.com     /  Emergency  \
       +----+   \                    |   Services    |
                 \  ,-------.        |   Network     |
                  ,'         `.      |               |
                 /   VoIP      \     |   +------+    |
                (    Provider   )    |   |PSAP  |    |
                 \             /     |   +--+---+    |
                  `.         ,'      |      |
                    '---+---'        |      |        |
                        |            |psap@town.com  |
                esrp@state.com       |      |        |
                        |            |      |        |
                        |            |      |        |
                        |            |   +--+---+    |
                        +------------+---+ESRP  |    |
                                     |   +------+    |
                                     |               |
                                      \             /
                                       `.         ,'
                                         '-------'

                 Figure 2: Example for Multi-Stage Routing

3.3.  Call Forwarding

   Imagine the following case where an emergency call enters an
   emergency network (state.org) via an ERSP but then gets forwarded to
   a different emergency services network (in our example to police-
   town.org, fire-town.org or medic-town.org).  The same considerations
   apply when the police, fire and ambulance networks are part of the
   state.org sub-domains (e.g., police.state.org).

   Similarly to the previous scenario the problem here is with the wrong
   state information being established during the emergency call setup
   procedure.  A callback would originate in the police-town.org, fire-
   town.org or medic-town.org domain whereas the emergency caller's SIP
   UA or the VoIP outbound proxy has stored state.org.








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                                   ,-------.
                                 ,'         `.
                                /  Emergency  \
                               |   Services    |
                               |   Network     |
                               |   (state.org) |
                               |               |
                               |               |
                               |   +------+    |
                               |   |PSAP  +--+ |
                               |   +--+---+  | |
                               |      |      | |
                               |      |      | |
                               |      |      | |
                               |      |      | |
                               |      |      | |
                               |   +--+---+  | |
             ------------------+---+ESRP  |  | |
             esrp-a@state.org  |   +------+  | |
                               |             | |
                               |    Call Fwd | |
                               |     +-+-+---+ |
                                \    | | |    /
                                 `.  | | |  ,'
                                   '-|-|-|-'           ,-------.
                            Police   | | | Fire      ,'         `.
                        +------------+ | +----+     /  Emergency  \
         ,-------.      |              |      |    |   Services    |
       ,'         `.    |              |      |    |   Network     |
      /  Emergency  \   |          Ambulance  |    | fire-town.org |
     |   Services    |  |              |      |    |               |
     |   Network     |  |              +----+ |    |   +------+    |
     |police-town.org|  |     ,-------.     | +----+---+PSAP  |    |
     |               |  |   ,'         `.   |      |   +------+    |
     |   +------+    |  |  /  Emergency  \  |      |               |
     |   |PSAP  +----+--+ |   Services    | |      |               ,
     |   +------+    |    |   Network     | |      `~~~~~~~~~~~~~~~
     |               |    |medic-town.org | |
     |               ,    |               | |
     `~~~~~~~~~~~~~~~     |   +------+    | |
                          |   |PSAP  +----+ +
                          |   +------+    |
                          |               |
                          |               ,
                          `~~~~~~~~~~~~~~~

                   Figure 3: Example for Call Forwarding




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3.4.  Network-based Service URN Resolution

   The IETF emergency services architecture also considers cases where
   the resolution from the Service URN to the PSAP URI does not only
   happen at the SIP UA itself but at intermediate SIP entities, such as
   the user's VoIP provider.

   Figure 4 shows this message exchange of the outgoing emergency call
   and the incoming PSAP graphically.  While the state information
   stored at the VoIP provider is correct the state allocated at the SIP
   UA is not.


        ,-------.
      ,'         `.
     /  Emergency  \
    |   Services    |
    |   Network     |
    |police-town.org|
    |               |
    |   +------+    |    Invite to police.example.com
    |   |PSAP  +<---+------------------------+
    |   |      +----+------------------+     ^
    |   +------+    |Invite from       |     |
    |               ,police.example.com|     |
    `~~~~~~~~~~~~~~~                   v     |
    +--------+                        ++-----+-+
    |        |            query       |VoIP    |
    | LoST   |<-----------------------|Service |
    | Server |   police.example.com   |Provider|
    |        |----------------------->|        |
    +--------+                        +--------+
                                       |     ^
                                 Invite|     | Invite
                                   from|     | to
                     police.example.com|     | urn:service:sos
                                       V     |
                                      +-------+
                                      | SIP   |
                                      | UA    |
                                      | Alice |
                                      +-------+

        Figure 4: Example for Network-based Service URN Resolution







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3.5.  PSTN Interworking

   In case an emergency call enters the PSTN, as shown in Figure 5,
   there is no guarantee that the callback some time later does leave
   the same PSTN/VoIP gateway or that the same end point identifier is
   used in the forward as well as in the backward direction making it
   difficult to reliably detect PSAP callbacks.


     +-----------+
     | PSTN      |-------------+
     | Calltaker |             |
     | Bob       |<--------+   |
     +-----------+         |   v
                -------------------
            ////                   \\\\      +------------+
           |                           |     |PSTN / VoIP |
           |             PSTN          |---->|Gateway     |
            \\\\                   ////      |            |
                -------------------          +----+-------+
                           ^                      |
                           |                      |
                     +-------------+              |  +--------+
                     |             |              |  |VoIP    |
                     | PSTN / VoIP |              +->|Service |
                     | Gateway     |                 |Provider|
                     |             |<------Invite----|   Y    |
                     +-------------+                 +--------+
                                                      |     ^
                                                      |     |
                                                    Invite Invite
                                                      |     |
                                                      V     |
                                                     +-------+
                                                     | SIP   |
                                                     | UA    |
                                                     | Alice |
                                                     +-------+

                  Figure 5: Example for PSTN Interworking

   Note: This scenario is considered outside the scope of this document.
   The specified solution does not support this use case.








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4.  SIP PSAP Callback Indicator

4.1.  General

   This section defines a new header field value, called "psap-
   callback", for the SIP Priority header field defined in [RFC3261].
   The value is used to inform SIP entities that the request is
   associated with a PSAP callback SIP session.

4.2.  Usage

   SIP entities that receive the header field value within an initial
   request for a SIP session can, depending on local policies, apply
   PSAP callback specific procedures for the session or request.

   The PSAP callback specific procedures may be applied by SIP-based
   network entities and by the callee.  The specific procedures taken
   when receiving such a PSAP callback marked call, such as bypassing
   services and barring procedures, are outside the scope of this
   document.

4.3.  Syntax

4.3.1.  General

   This section defines the ABNF for the new SIP Priority header field
   value "psap-callback".

4.3.2.  ABNF


           priority-value  /=  "psap-callback"

                              Figure 6: ABNF

















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5.  Security Considerations

5.1.  Security Threat

   The PSAP callback functionality described in this document allows
   marked calls to bypass blacklists, ignore call forwarding procedures
   and similar features to contact emergency callers and to raise their
   attention.  Regarding the latter aspect a callback, if understood by
   the SIP UA would allow to override user interface configurations,
   such as vibrate-only mode, to alert the caller of the incoming call.

5.2.  Security Requirements

   The requirement is to ensure that the mechanisms described in this
   document can not be used for malicious purposes, including
   telemarketing.

   Furthermore, if the newly defined extension is not recognized, not
   verified adequately, or not obeyed by SIP intermediaries or SIP
   endpoints then it must not lead to a failure of the call handling
   procedure.  Such call must be treated like a call that does not have
   any marking attached.

5.3.  Security Solution

   Figure 7 shows the architecture that utilizes the identity of the
   PSAP to decide whether a preferential treatment of callbacks should
   be provided.  To make this policy decision the identity of the PSAP
   is compared with a whitelist of valid PSAPs available to the SIP
   entity.  The identity assurance in SIP can come in different forms,
   such as SIP Identity [RFC4474] or with P-Asserted-Identity [RFC3325].
   The former technique relies on a cryptographic assurance and the
   latter on a chain of trust.  Also the usage of TLS between
   neighboring SIP entities may provide useful identity information.

















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                       +----------+
                       | List of  |+
                       | valid    ||
                       | PSAPs    ||
                       +----------+|
                        +----------+
                            *
                            * whitelist
                            *
                            V
         Incoming      +----------+    Normal
         SIP Msg       | SIP      |+   Treatment
        -------------->| Entity   ||======================>
         + Identity    |          ||(if not in whitelist)
           Info        +----------+|
                       +----------+
                            ||
                            ||
                            || Preferential
                            || Treatment
                            ++========================>
                              (if successfully verified)

                  Figure 7: Identity-based Authorization

   An important aspect from a security point of view is the relationship
   between the emergency services network (containing PSAPs) and the VSP
   (assuming that the emergency call travels via the VSP and not
   directly between the SIP UA and the PSAP).

   If there is some form of relationship between the emergency services
   operator and the VSP then the identification of a PSAP call back is
   less problematic than in the case where the two entities have not
   entered in some form of relationship that would allow the VSP to
   verify whether the marked callback message indeed came from a
   legitimate source.

   The establishment of a whitelist with PSAP identities maybe be
   operationally complex.  When there is a local relationship between
   the VSP/ASP and the PSAP then populating the whitelist is fairly
   simple.  For SIP UAs there is no need to maintain a list of PSAPs.
   Instead SIP UAs are assumed to trust the correct processing of their
   VSP/ASP, i.e., the VSP/ASP processes the PSAP callback marking and,
   if it cannot be verified, the PSAP callback marking is removed.  If
   it is left untouched then the SIP UA should assume that it has been
   verified successfully by the VSP/ASP and it should therefore be
   obeyed.




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6.  IANA Considerations

   [Editor's Note: There is currently no registry available for the SIP
   Priority header.  A registry needs to be created and the value
   defined in this document needs to be added.]














































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7.  Acknowledgements

   We would like to thank members from the ECRIT working group, in
   particular Brian Rosen, for their discussions around PSAP callbacks.
   The working group discussed the topic of callbacks at their virtual
   interim meeting in February 2010 and the following persons provided
   valuable input: John Elwell, Bernard Aboba, Cullen Jennings, Keith
   Drage, Marc Linsner, Roger Marshall, Dan Romascanu, Geoff Thompson,
   Janet Gunn.

   At IETF#81 a small group of people got to together to continue the
   discussions started at the working group meeting to explore a GRUU-
   based solution approach.  Martin Thomson, Marc Linsner, Andrew Allen,
   Brian Rosen, Martin Dolly, and Atle Monrad participated at this side-
   meeting.

   We would like to thank the following persons for their feedback on
   the solution discussion in 2012: Paul Kyzivat, Martin Thomson, Robert
   Sparks, Keith Drage, Brian Rosen, Roger Marshall, Martin Dolly,
   Bernard Aboba, Andrew Allen, John-Luc Bakker, James Polk, John
   Medland, Hadriel Kaplan, Kenneth Carlberg, Timothy Dwight, Janet Gunn






























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8.  References

8.1.  Normative References

   [RFC2119]                  Bradner, S., "Key words for use in RFCs to
                              Indicate Requirement Levels", BCP 14,
                              RFC 2119, March 1997.

   [RFC3261]                  Rosenberg, J., Schulzrinne, H., Camarillo,
                              G., Johnston, A., Peterson, J., Sparks,
                              R., Handley, M., and E. Schooler, "SIP:
                              Session Initiation Protocol", RFC 3261,
                              June 2002.

   [RFC3325]                  Jennings, C., Peterson, J., and M. Watson,
                              "Private Extensions to the Session
                              Initiation Protocol (SIP) for Asserted
                              Identity within Trusted Networks",
                              RFC 3325, November 2002.

   [RFC3966]                  Schulzrinne, H., "The tel URI for
                              Telephone Numbers", RFC 3966,
                              December 2004.

   [RFC3969]                  Camarillo, G., "The Internet Assigned
                              Number Authority (IANA) Uniform Resource
                              Identifier (URI) Parameter Registry for
                              the Session Initiation Protocol (SIP)",
                              BCP 99, RFC 3969, December 2004.

   [RFC4474]                  Peterson, J. and C. Jennings,
                              "Enhancements for Authenticated Identity
                              Management in the Session Initiation
                              Protocol (SIP)", RFC 4474, August 2006.

   [RFC5627]                  Rosenberg, J., "Obtaining and Using
                              Globally Routable User Agent URIs (GRUUs)
                              in the Session Initiation Protocol (SIP)",
                              RFC 5627, October 2009.

8.2.  Informative References

   [I-D.ietf-ecrit-phonebcp]  Rosen, B. and J. Polk, "Best Current
                              Practice for Communications Services in
                              support of Emergency Calling",
                              draft-ietf-ecrit-phonebcp-20 (work in
                              progress), September 2011.




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   [RFC4484]                  Peterson, J., Polk, J., Sicker, D., and H.
                              Tschofenig, "Trait-Based Authorization
                              Requirements for the Session Initiation
                              Protocol (SIP)", RFC 4484, August 2006.

   [RFC5012]                  Schulzrinne, H. and R. Marshall,
                              "Requirements for Emergency Context
                              Resolution with Internet Technologies",
                              RFC 5012, January 2008.

   [RFC5031]                  Schulzrinne, H., "A Uniform Resource Name
                              (URN) for Emergency and Other Well-Known
                              Services", RFC 5031, January 2008.

   [RFC5234]                  Crocker, D. and P. Overell, "Augmented BNF
                              for Syntax Specifications: ABNF", STD 68,
                              RFC 5234, January 2008.

   [RFC6444]                  Schulzrinne, H., Liess, L., Tschofenig,
                              H., Stark, B., and A. Kuett, "Location
                              Hiding: Problem Statement and
                              Requirements", RFC 6444, January 2012.





























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Authors' Addresses

   Henning Schulzrinne
   Columbia University
   Department of Computer Science
   450 Computer Science Building
   New York, NY  10027
   US

   Phone: +1 212 939 7004
   EMail: hgs+ecrit@cs.columbia.edu
   URI:   http://www.cs.columbia.edu


   Hannes Tschofenig
   Nokia Siemens Networks
   Linnoitustie 6
   Espoo  02600
   Finland

   Phone: +358 (50) 4871445
   EMail: Hannes.Tschofenig@gmx.net
   URI:   http://www.tschofenig.priv.at


   Christer Holmberg
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   EMail: christer.holmberg@ericsson.com


   Milan Patel
   InterDigital Communications


   EMail: Milan.Patel@interdigital.com












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