[Docs] [txt|pdf] [draft-ietf-ecrit-...] [Diff1] [Diff2] [IPR]


Internet Engineering Task Force (IETF)                    H. Schulzrinne
Request for Comments: 7090                           Columbia University
Category: Standards Track                                  H. Tschofenig
ISSN: 2070-1721
                                                             C. Holmberg
                                                                M. Patel
                                      Huawei Technologies (UK) Co., Ltd.
                                                              April 2014

             Public Safety Answering Point (PSAP) Callback


   After an emergency call is completed (terminated either prematurely
   by the emergency caller or normally by the call taker), the call
   taker may feel 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 state of an accident victim.
   A call taker may trigger a callback to 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 Public Safety Answering Point (PSAP)
   callbacks to bypass authorization policies in order 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.  We describe a solution based on a new header field value
   for the SIP Priority header field, called "psap-callback", to mark
   PSAP callbacks.

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Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at

Copyright Notice

   Copyright (c) 2014 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 ....................................................3
   2. Terminology .....................................................5
   3. Callback Scenarios ..............................................5
      3.1. Routing Asymmetry ..........................................5
      3.2. Multi-Stage Routing ........................................7
      3.3. Call Forwarding ............................................8
      3.4. Network-Based Service URN Resolution ......................10
      3.5. PSTN Interworking .........................................11
   4. SIP PSAP Callback Indicator ....................................12
      4.1. General ...................................................12
      4.2. Usage .....................................................12
      4.3. Syntax ....................................................12
           4.3.1. General ............................................12
           4.3.2. ABNF ...............................................12
   5. Security Considerations ........................................12
      5.1. Security Threat ...........................................12
      5.2. Security Requirements .....................................13
      5.3. Security Solution .........................................13
   6. IANA Considerations ............................................15
   7. Acknowledgements ...............................................16
   8. References .....................................................16
      8.1. Normative References ......................................16
      8.2. Informative References ....................................17

1.  Introduction

   Summoning police, the fire department, or an ambulance in emergencies
   is one of the fundamental and most valuable 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 and that
   are not traditional telephones.  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 is described in [RFC6443],
   and [RFC6881] 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 (GRUUs).  RFC 3261 [RFC3261] defines the AOR as:

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      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.) that are all
   referenced by the same AOR.  There are a number of cases in which it
   is desirable to have an identifier that 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, and it is also globally
   routable.  [RFC5627] specifies how to obtain and use GRUUs.
   [RFC6881] 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 callback to the emergency caller.  This is desirable in
   those cases where the call is dropped prematurely or when further
   communication needs arise.  The AOR and the GRUU serve this purpose.

   The communication attempt by the PSAP call taker back to the
   emergency caller is called a "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
   itself) cannot differentiate the PSAP callback from any other SIP
   call.  "Call barring", "do not disturb", or "call diversion" (also
   called 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 a desire to treat
   PSAP callbacks in such a way that the chances of reaching the
   emergency caller are increased.  At the same time, any solution must
   minimize the chance that other calls 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, but [RFC6881]
   suggests 30 minutes.

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   Nevertheless, to meet the needs from the emergency services
   community, a basic mechanism for preferential treatment of PSAP
   callbacks was defined in Section 13 of [RFC6443].  The specification

      A UA may be able to determine a PSAP callback 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 call 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

2.  Terminology

   Emergency-services-related terminology is borrowed from [RFC5012].
   This includes terminology like emergency caller, user equipment, call
   taker, Emergency Service Routing Proxy (ESRP), and Public Safety
   Answering Point (PSAP).

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

3.  Callback Scenarios

   This section illustrates a number of scenarios where the currently
   specified solution, as described in [RFC6881], for preferential
   treatment of callbacks fails.  As explained in Section 1, a SIP
   entity examines an incoming PSAP callback by comparing the domain of
   the PSAP with the destination domain of the outbound emergency call
   placed earlier.

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 Voice over IP (VoIP)
   provider uses different SIP proxies for inbound and for outbound call
   handling.  Unless the two devices are synchronized, the callback

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   reaching the inbound proxy would get treated like any other call
   since the emergency call established state information at the
   outbound proxy only.

                                                 ,'         `.
                      ,-------.                 /  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

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3.2.  Multi-Stage Routing

   Consider the 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 the Location-to-Service Translation
   (LoST) Protocol [RFC5222] to learn the next-hop destination, namely
   esrp@example.net, to get the call closer to the PSAP.  This call is
   then sent to the proxy of the user's VoIP provider (example.org).
   The user's VoIP provider receives the emergency call and creates a
   state based on the destination domain, namely example.net.  It then
   routes the call to the indicated ESRP.  When the ESRP receives the
   call, it needs to decide what the next hop is to get to the final
   PSAP.  In our example, the next hop is the PSAP with the URI

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

       +----+                          ,'             `.
       | UA |--- esrp@example.net    /     Emergency    \
       +----+   \                    |      Services    |
                 \  ,-------.        |      Network     |
                  ,'         `.      |                  |
                 /   VoIP      \     |     +------+     |
                (   Provider    )    |     | PSAP |     |
                 \ example.org /     |     +--+---+     |
                  `.         ,'      |        |         |
                    '---+---'        |        |         |
                        |            | psap@example.com |
                esrp@example.net     |        |         |
                        |            |        |         |
                        |            |        |         |
                        |            |     +--+---+     |
                        +------------+-----+ ESRP |     |
                                     |     +------+     |
                                     |                  |
                                      \                /
                                       `.            ,'

                 Figure 2: Example for Multi-Stage Routing

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3.3.  Call Forwarding

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

   Similar to the previous scenario, the wrong state information is
   being set up during the emergency call setup procedure.  A callback
   would originate in the example.net, example.org, or example.com
   domains whereas the emergency caller's SIP UA or the VoIP outbound
   proxy has stored state.example.

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                                 ,'         `.
                                /  Emergency  \
                               |   Services    |
                               |   Network     |
                               |               |
                               |               |
                               |   +------+    |
                               |   |PSAP  +--+ |
                               |   +--+---+  | |
                               |      |      | |
                               |      |      | |
                               |      |      | |
                               |      |      | |
                               |      |      | |
                               |   +--+---+  | |
             ------------------+---+ESRP  |  | |
             esrp-a@state.org  |   +------+  | |
                               |             | |
                               |    Call Fwd | |
                               |     +-+-+---+ |
                                \    | | |    /
                                 `.  | | |  ,'
                                   '-|-|-|-'           ,-------.
                            Police   | | | Fire      ,'         `.
                        +------------+ | +----+     /  Emergency  \
         ,-------.      |              |      |    |   Services    |
       ,'         `.    |              |      |    |   Network     |
      /  Emergency  \   |          Ambulance  |    |    (Fire)     |
     |   Services    |  |              |      |    |               |
     |   Network     |  |              +----+ |    |   +------+    |
     |   (Police)    |  |     ,-------.     | +----+---+PSAP  |    |
     |               |  |   ,'         `.   |      |   +------+    |
     |   +------+    |  |  /  Emergency  \  |      |               |
     |   |PSAP  +----+--+ |   Services    | |      |  example.com  ,
     |   +------+    |    |   Network     | |      `~~~~~~~~~~~~~~~
     |               |    |  (Ambulance)  | |
     |  example.net  ,    |               | |
     `~~~~~~~~~~~~~~~     |   +------+    | |
                          |   |PSAP  +----+ +
                          |   +------+    |
                          |               |
                          |  example.org  ,

                   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     |
    |  example.com  |
    |               |
    |   +------+    |    INVITE to police@example.com
    |   |PSAP  +<---+------------------------+
    |   |      +----+--------------------+   ^
    |   +------+    |INVITE from         |   |
    |               ,police@example.com  |   |
    `~~~~~~~~~~~~~~~                     |   |
                                         v   |
    +--------+  Query with location   +--+---+-+
    |        |  + urn:service:sos     |  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 Public Switched Telephone
   Network (PSTN), as shown in Figure 5, there is no guarantee that the
   callback sometime later leaves the same PSTN/VoIP gateway or that the
   same endpoint identifier is used in the forward as well as in the
   backward direction making it difficult to reliably detect PSAP

     | 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 actions taken when
   receiving a call marked as 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 [RFC5234] for the new SIP Priority
   header field value "psap-callback".

4.3.2.  ABNF

       priority-value  =/  "psap-callback"

               Figure 6: ABNF

5.  Security Considerations

5.1.  Security Threat

   The PSAP callback functionality described in this document allows
   marked calls to bypass blacklists and ignore call-forwarding
   procedures and other similar features used to raise the attention of
   emergency callers when attempting to contact them.  In the case where
   the SIP Priority header value, "psap-callback", is supported by the
   SIP UA, it would override user-interface configurations, such as
   vibrate-only mode, to alert the caller of the incoming call.

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5.2.  Security Requirements

   The security threat discussed in Section 5.1 leads to the requirement
   to ensure that the mechanisms described in this document cannot 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 a call must be treated like a call that does not
   have any marking attached.

   The indicator described in Section 4 can be inserted by any SIP
   entity, including attackers.  So it is critical that the indicator
   only lead to preferential call treatment in cases where the recipient
   has some trust in the caller, as described in the next section.

5.3.  Security Solution

   The approach for dealing with the implementation of the security
   requirements described in Section 5.2 can be differentiated between
   the behavior applied by the UA and by SIP proxies.  A UA that has
   made an emergency call MUST keep state information so that it can
   recognize and accept a callback from the PSAP if it occurs within a
   reasonable time after an emergency call was placed, as described in
   Section 13 of [RFC6443].  Only a timer started at the time when the
   original emergency call has ended is required; information about the
   calling party identity is not needed since the callback may use a
   different calling party identity, as described in Section 3.  Since
   these SIP UA considerations are described already in [RFC6443] as
   well as in [RFC6881] the rest of this section focuses on the behavior
   of SIP proxies.

   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
   (i.e., calling party identity) is compared with a PSAPs white list.

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

                  Figure 7: Identity-Based Authorization

   The identity assurance in SIP can come in different forms, namely via
   the SIP Identity [RFC4474] or the P-Asserted-Identity [RFC3325]
   mechanisms.  The former technique relies on a cryptographic assurance
   and the latter on a chain of trust.  Also, the usage of Transport
   Layer Security (TLS) between neighboring SIP entities may provide
   useful identity information.  At the time of writing, these identity
   technologies are being revised in the Secure Telephone Identity
   Revisited (stir) working group [STIR] to offer better support for
   legacy technologies interworking and SIP intermediaries that modify
   the content of various SIP headers and the body.  Once the work on
   these specifications has been completed, they will offer a stronger
   calling party identity mechanism that limits or prevents identity

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

   The establishment of a white list with PSAP identities may be
   operationally complex and dependent on the relationship between the
   emergency services operator and the VoIP provider.  If there is a
   relationship between the VoIP provider and the PSAP operator, for

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   example, when they are both operating in the same geographical
   region, then populating the white list is fairly simple and
   consequently the identification of a PSAP callback is less
   problematic compared to the case where the two entities have never
   interacted with each other before.  In the end, the VoIP provider has
   to verify whether the marked callback message indeed came from a
   legitimate source.

   VoIP providers MUST only give PSAP callbacks preferential treatment
   when the calling party identity of the PSAP was successfully matched
   against entries in the white list.  If it cannot be verified (because
   there was no match), then the VoIP provider MUST remove the PSAP
   callback marking.  Thereby, the callback reverts to a normal call.
   As a second step, SIP UAs MUST maintain a timer that is started with
   the original emergency call and this timer expires within a
   reasonable amount of time, such as 30 minutes per [RFC6881].  Such a
   timer also ensures that VoIP providers cannot misuse the PSAP
   callback mechanism, for example, to ensure that their support calls
   reach their customers.

   Finally, a PSAP callback MUST use the same media as the original
   emergency call.  For example, when an initial emergency call
   established a real-time text communication session, then the PSAP
   callback must also attempt to establish a real-time communication
   interaction.  The reason for this is twofold.  First, the person
   seeking help may have disabilities that prevent them from using
   certain media and hence using the same media for the callback avoids
   unpleasant surprises and delays.  Second, the emergency caller may
   have intentionally chosen a certain media and does not prefer to
   communicate in a different way.  For example, it would be unfortunate
   if a hostage tries to seek help using instant messaging to avoid any
   noise when subsequently the ringtone triggered by a PSAP callback
   using a voice call gets the attention of the hostage-taker.  User-
   interface designs need to cater to such situations.

6.  IANA Considerations

   This document adds the "psap-callback" value to the SIP "Priority
   Header Field Values" registry allocated by [RFC6878].  The semantic
   of the newly defined "psap-callback" value is defined in Section 4.

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7.  Acknowledgements

   We would like to thank the following persons for their feedback:
   Bernard Aboba, Andrew Allen, John-Luc Bakker, Kenneth Carlberg,
   Martin Dolly, Keith Drage, Timothy Dwight, John Elwell, Janet Gunn,
   Cullen Jennings, Hadriel Kaplan, Paul Kyzivat, John Medland, Atle
   Monrad, James Polk, Dan Romascanu, Brian Rosen, Robert Sparks, Geoff
   Thompson, and Martin Thomson.

   We would also like to thank the ECRIT working group chairs, Marc
   Linsner and Roger Marshall, for their support.  Roger Marshall was
   the document shepherd for this document.  Vijay Gurbani provided the
   general area review.

   During IESG review, the document received good feedback from Barry
   Leiba, Spencer Dawkins, Richard Barnes, Joel Jaeggli, Stephen
   Farrell, and Benoit Claise.

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.

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

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

   [RFC6878]  Roach, A., "IANA Registry for the Session Initiation
              Protocol (SIP) "Priority" Header Field", RFC 6878, March

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8.2.  Informative References

   [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.

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

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

   [RFC5222]  Hardie, T., Newton, A., Schulzrinne, H., and H.
              Tschofenig, "LoST: A Location-to-Service Translation
              Protocol", RFC 5222, August 2008.

   [RFC6443]  Rosen, B., Schulzrinne, H., Polk, J., and A. Newton,
              "Framework for Emergency Calling Using Internet
              Multimedia", RFC 6443, December 2011.

   [RFC6881]  Rosen, B. and J. Polk, "Best Current Practice for
              Communications Services in Support of Emergency Calling",
              BCP 181, RFC 6881, March 2013.

   [STIR]     IETF, "Secure Telephone Identity Revisited (stir) Working
              Group", http://datatracker.ietf.org/wg/stir/charter/,
              October 2013.

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Authors' Addresses

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

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

   Hannes Tschofenig

   EMail: Hannes.Tschofenig@gmx.net
   URI:   http://www.tschofenig.priv.at

   Christer Holmberg
   Hirsalantie 11
   Jorvas  02420

   EMail: christer.holmberg@ericsson.com

   Milan Patel
   Huawei Technologies (UK) Co., Ltd.
   300 South Oak Way, Green Park
   Reading, Berkshire  RG2 6UF

   EMail: Milan.Patel@huawei.com

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