One document matched: draft-ietf-ecrit-psap-callback-13.xml
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]>
<rfc category="std" docName="draft-ietf-ecrit-psap-callback-13.txt"
ipr="trust200902">
<?rfc toc="yes" ?>
<?rfc symrefs="yes" ?>
<?rfc sortrefs="yes"?>
<?rfc rfcedstyle="yes" ?>
<?rfc subcompact="no"?>
<?rfc compact="no"?>
<?rfc strict="no"?>
<front>
<title abbrev="PSAP Callback">Public Safety Answering Point (PSAP)
Callback</title>
<author fullname="Henning Schulzrinne" initials="H." surname="Schulzrinne">
<organization>Columbia University</organization>
<address>
<postal>
<street>Department of Computer Science</street>
<street>450 Computer Science Building</street>
<city>New York</city>
<region>NY</region>
<code>10027</code>
<country>US</country>
</postal>
<phone>+1 212 939 7004</phone>
<email>hgs+ecrit@cs.columbia.edu</email>
<uri>http://www.cs.columbia.edu</uri>
</address>
</author>
<author fullname="Hannes Tschofenig" initials="H." surname="Tschofenig">
<organization>Nokia Solutions and Networks</organization>
<address>
<postal>
<street>Linnoitustie 6</street>
<city>Espoo</city>
<code>02600</code>
<country>Finland</country>
</postal>
<phone>+358 (50) 4871445</phone>
<email>Hannes.Tschofenig@gmx.net</email>
<uri>http://www.tschofenig.priv.at</uri>
</address>
</author>
<author initials="C.H." surname="Holmberg" fullname="Christer Holmberg">
<organization>Ericsson</organization>
<address>
<postal>
<street>Hirsalantie 11</street>
<code>02420</code>
<city>Jorvas</city>
<country>Finland</country>
</postal>
<email>christer.holmberg@ericsson.com</email>
</address>
</author>
<author fullname="Milan Patel" initials="M." surname="Patel">
<organization>InterDigital Communications</organization>
<address>
<postal>
<street></street>
<city></city>
<code></code>
<country></country>
</postal>
<email>Milan.Patel@interdigital.com</email>
</address>
</author>
<date year="2013" />
<workgroup>ECRIT</workgroup>
<abstract>
<t>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.</t>
<t>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. A solution based on a new header field value, called "psap-callback", for the SIP Priority header
field is specified to accomplish the PSAP callback marking.</t>
</abstract>
</front>
<middle>
<!-- ****************************************************************************************** -->
<section anchor="intro" title="Introduction">
<t>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.</t>
<t>An overview of the protocol interactions for emergency calling using the IETF emergency services architecture are described in <xref target="RFC6443"/> and <xref target="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 (GRUU). RFC 3261 <xref target="RFC3261"/> defines the AOR as:
<list style="empty">
<t>
"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."
</t>
</list>
</t>
<t>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 <xref target="RFC5627"/> specifies how to obtain and use GRUUs. <xref target="RFC6881"/> also makes use of the GRUU for emergency calls.
</t>
<t>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 got dropped
prematurely or when further communication need arises.
The AOR and the GRUU serve this purpose.</t>
<t>The communication attempt by the PSAP call taker back to the emergency caller is called 'PSAP callback'.
</t>
<t>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"(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. </t>
<t>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 but <xref target="RFC6881"/> suggests 30 minutes.</t>
<t>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 <xref
target="RFC6443"></xref>. The specification says:</t>
<t><list style="empty">
<t>"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."</t>
</list></t>
<t>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 <xref target="scenarios"/> we describe cases where the currently
standardized approach is insufficient.</t>
<t><!-- In <xref target="solution"/> a solution is described. -->
</t>
</section>
<!-- ****************************************************************************************** -->
<section anchor="terminology" title="Terminology">
<!-- <t>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 <xref
target="RFC2119"></xref>.</t>
-->
<t>Emergency services related terminology is borrowed from <xref
target="RFC5012"></xref>. This includes terminology like emergency caller, user equipment,
call taker, Emergency Service Routing Proxy (ESRP), and Public Safety Answering Point (PSAP).
</t>
</section>
<!-- ****************************************************************************************** -->
<section anchor="scenarios" title="Callback Scenarios">
<t>This section illustrates a number of scenarios where the currently specified solution, as specified in <xref target="RFC6881"/>, for preferential treatment of callbacks fails. As explained in <xref target="intro"/> 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.</t>
<!-- <t>NOTE: All FQDNs used in the subsections below are used for illustrative purposes. They are examples to demonstrate the limitations of the technical solution outlined in RFC 6881.</t> -->
<section title="Routing Asymmetry">
<t>In some deployment environments it is common to have incoming and
outgoing SIP messaging routed through different SIP entities. <xref target="asymmetry"/> shows this graphically whereby a VoIP provider uses different SIP proxies for inbound and for outbound call handling. Unless the two devices are 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. </t>
<t><figure anchor="asymmetry" title="Example for Routing Asymmetry.">
<artwork>
<![CDATA[
,-------.
,' `.
,-------. / 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 |+-+ |
\ / | |
`. ,' \ /
'-------' `. ,'
'-------'
]]></artwork>
</figure></t>
</section>
<section anchor="multistage" title="Multi-Stage Routing">
<t>Consider the following emergency call routing scenario shown in
<xref target="stages"></xref> where routing towards the PSAP occurs in
several stages. In this scenario we consider a SIP UA that uses the Location-to-Service Translation Protocol (LoST) <xref target="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 state based on the destination domain, namely example.net. It then routes it to the indicated ESRP.
When the ESRP receives it 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.</t>
<t>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.</t>
<t><figure anchor="stages" title="Example for Multi-Stage Routing.">
<artwork><![CDATA[
,-----------.
+----+ ,' `.
| UA |--- esrp@example.net / Emergency \
+----+ \ | Services |
\ ,-------. | Network |
,' `. | |
/ VoIP \ | +------+ |
( Provider ) | | PSAP | |
\ example.org / | +--+---+ |
`. ,' | | |
'---+---' | | |
| | psap@example.com |
esrp@example.net | | |
| | | |
| | | |
| | +--+---+ |
+------------+-----+ ESRP | |
| +------+ |
| |
\ /
`. ,'
'----------'
]]></artwork>
</figure></t>
</section>
<section title="Call Forwarding">
<t>Imagine the following case where an emergency call enters an
emergency network (state.example) via an ESRP but then 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 sub-domains (e.g., police.state.example).</t>
<t>Similar 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 example.net, example.org or example.com
domains whereas the emergency caller's SIP UA or the VoIP outbound proxy has
stored state.example.</t>
<t><figure anchor="fwd" title="Example for Call Forwarding.">
<artwork><![CDATA[
,-------.
,' `.
/ 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 ,
`~~~~~~~~~~~~~~~
]]></artwork>
</figure></t>
</section>
<section title="Network-based Service URN Resolution">
<t>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.</t>
<t><xref target="late-binding"></xref> 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. </t>
<t><figure anchor="late-binding"
title="Example for Network-based Service URN Resolution.">
<artwork><![CDATA[
,-------.
,' `.
/ 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 |
+-------+
]]></artwork>
</figure></t>
</section>
<section title="PSTN Interworking">
<t>In case an emergency call enters the PSTN, as shown in <xref
target="pstn"></xref>, there is no guarantee that the callback some
time later leaves 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.</t>
<t><figure anchor="pstn" title="Example for PSTN Interworking.">
<artwork><![CDATA[
+-----------+
| PSTN |-------------+
| Calltaker | |
| Bob |<--------+ |
+-----------+ | v
-------------------
//// \\\\ +------------+
| | |PSTN / VoIP |
| PSTN |---->|Gateway |
\\\\ //// | |
------------------- +----+-------+
^ |
| |
+-------------+ | +--------+
| | | |VoIP |
| PSTN / VoIP | +->|Service |
| Gateway | |Provider|
| |<------Invite----| Y |
+-------------+ +--------+
| ^
| |
Invite Invite
| |
V |
+-------+
| SIP |
| UA |
| Alice |
+-------+
]]></artwork>
</figure></t>
<t>Note: This scenario is considered outside the scope of this
document. The specified solution does not support this use case.</t>
</section>
</section>
<!-- ****************************************************************************************** -->
<section anchor="solution" title="SIP PSAP Callback Indicator">
<section title="General">
<t>
This section defines a new header field value, called "psap-callback", for the SIP Priority header
field defined in <xref target="RFC3261"/>. The value is used to inform
SIP entities that the request is associated with a PSAP callback SIP session.
</t>
</section>
<section title="Usage">
<t>
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.
</t>
<t>
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.
</t>
</section>
<section title="Syntax">
<section title="General">
<t>
This section defines the ABNF for the new SIP Priority header field value "psap-callback".
</t>
</section>
<section title="ABNF">
<t><figure anchor="abnf" title="ABNF">
<artwork><![CDATA[
priority-value /= "psap-callback"
]]></artwork>
</figure></t>
</section>
</section>
</section>
<!-- ****************************************************************************************** -->
<section title="Security Considerations">
<section anchor="threat" title="Security Threat">
<t>
The PSAP callback functionality described in this document allows
marked calls to bypass blacklists, 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.
</t>
</section>
<section anchor="req" title="Security Requirements">
<t>The security threat discussed in <xref target="threat"/> leads
to the requirement to ensure that the
mechanisms described in this document can not be used for
malicious purposes, including telemarketing.
</t>
<t>
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.
</t>
<t>The indicator described in <xref target="solution"/> 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.</t>
</section>
<section title="Security Solution">
<t>The approach for dealing with implementing the security requirements described in <xref target="req"/>
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 accepted a callback from the PSAP if it occurs within a
reasonable time after an emergency call was placed, as described in Section 13 of <xref target="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 <xref target="scenarios"/>. Since these SIP UA considerations are described already in <xref target="RFC6443"/> as well as in <xref target="RFC6881"/> the rest of this section focuses on the behavior of SIP proxies.</t>
<t><xref target="identity-authz"/> 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.
<figure anchor="identity-authz" title="Identity-based Authorization">
<artwork><![CDATA[
+----------+
| 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)
]]></artwork>
</figure></t>
<t>The identity assurance in SIP can come in different forms,
namely via the SIP Identity <xref target="RFC4474"/> or the P-Asserted-Identity <xref target="RFC3325"/> mechanisms.
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. At the time of writing these identity technologies are being revised in the Secure Telephone Identity Revisited (stir) working group <xref target="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.</t>
<t>
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).
</t>
<t>
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. When there is a relationship between
the VoIP provider and the PSAP operator, for 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.
</t>
<t>
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 is degenerated 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 <xref target="RFC6881"/>. Such a timer also ensures that VoIP providers cannot misuse the PSAP callback mechanism, for example to ensure that their support calls reaches their customers.
</t>
<t>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 two-fold. First, the person seeking for 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 for help using instant messaging to avoid any noise when subsequently the ring-tone 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.</t>
</section>
</section>
<!-- ****************************************************************************************** -->
<section anchor="iana" title="IANA Considerations">
<t>This document adds the "psap-callback" value to the SIP Priority header IANA registry allocated by <xref target="RFC6878"/>. The semantic of the newly defined "psap-callback" value is defined in <xref target="solution"/>.</t>
</section>
<!-- ****************************************************************************************** -->
<section title="Acknowledgements">
<!-- <t>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.</t>
<t>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.</t>
-->
<t>We would like to thank the following persons for their feedback:
Paul Kyzivat,
Martin Thomson,
Robert Sparks,
Keith Drage,
Cullen Jennings
Brian Rosen,
Martin Dolly,
Bernard Aboba,
Andrew Allen,
Atle Monrad,
John-Luc Bakker,
John Elwell,
Geoff Thompson,
Dan Romascanu,
James Polk,
John Medland,
Hadriel Kaplan,
Kenneth Carlberg,
Timothy Dwight,
Janet Gunn</t>
<t>We would 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.</t>
<t>During IESG review the document received good feedback from Barry Leiba, Spencer Dawkins, Richard Barnes, Joel Jaeggli, Stephen Farrell, and Benoit Claise.</t>
</section>
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<references title="Normative References">
&RFC3261;
&RFC5627;
&RFC6878;
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</references>
<references title="Informative References">
&RFC3325;
&RFC4474;
&RFC6443;
&RFC6881;
&RFC5012;
&RFC5222;
<!-- &RFC4484; -->
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<reference anchor="STIR">
<front>
<title>Secure Telephone Identity Revisited (stir) Working Group</title>
<author>
<organization>IETF</organization>
</author>
<date month="Oct" year="2013" />
</front>
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value="URL: http://datatracker.ietf.org/wg/stir/charter/" />
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| PAFTECH AB 2003-2026 | 2026-04-22 03:10:29 |