Public Safety Answering Point (PSAP) Callback
RFC 7090
| Document | Type | RFC - Proposed Standard (April 2014) IPR | |
|---|---|---|---|
| Authors | Henning Schulzrinne , Hannes Tschofenig , Christer Holmberg , Milan Patel | ||
| Last updated | 2018-12-20 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Additional resources | Mailing list discussion | ||
| IESG | Responsible AD | Richard Barnes | |
| Send notices to | (None) |
RFC 7090
Internet Engineering Task Force (IETF) H. Schulzrinne
Request for Comments: 7090 Columbia University
Category: Standards Track H. Tschofenig
ISSN: 2070-1721
C. Holmberg
Ericsson
M. Patel
Huawei Technologies (UK) Co., Ltd.
April 2014
Public Safety Answering Point (PSAP) Callback
Abstract
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
http://www.rfc-editor.org/info/rfc7090.
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.
Schulzrinne, et al. Standards Track [Page 2]
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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
says:
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
insufficient.
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",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
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
psap@example.com.
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
information.
,-----------.
+----+ ,' `.
| 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.,
police.state.example).
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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RFC 7090 PSAP Callback April 2014
,-------.
,' `.
/ Emergency \
| Services |
| Network |
|(state.example)|
| |
| |
| +------+ |
| |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
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 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
*
V
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
spoofing.
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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RFC 7090 PSAP Callback April 2014
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
2008.
[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
2013.
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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
US
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
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
EMail: christer.holmberg@ericsson.com
Milan Patel
Huawei Technologies (UK) Co., Ltd.
300 South Oak Way, Green Park
Reading, Berkshire RG2 6UF
U.K.
EMail: Milan.Patel@huawei.com
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