Public Safety Answering Point (PSAP) Callback
draft-ietf-ecrit-psap-callback-07
The information below is for an old version of the document.
| Document | Type | Active Internet-Draft (ecrit WG) | |
|---|---|---|---|
| Authors | Henning Schulzrinne , Hannes Tschofenig , Christer Holmberg , Milan Patel | ||
| Last updated | 2012-12-18 | ||
| Replaces | draft-schulzrinne-ecrit-psap-callback | ||
| Stream | Internet Engineering Task Force (IETF) | ||
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draft-ietf-ecrit-psap-callback-07
ECRIT H. Schulzrinne
Internet-Draft Columbia University
Intended status: Standards Track H. Tschofenig
Expires: June 21, 2013 Nokia Siemens Networks
C. Holmberg
Ericsson
M. Patel
InterDigital Communications
December 18, 2012
Public Safety Answering Point (PSAP) Callback
draft-ietf-ecrit-psap-callback-07.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 June 21, 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 . . . . . . . . . . . . . . . . . . 19
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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
This document adds the "psap-callback" value to the SIP Priority
header IANA registry allocated by [I-D.ietf-sipcore-priority]. 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 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
[I-D.ietf-sipcore-priority] Roach, A., "IANA Registry for the
Session Initiation Protocol (SIP)
"Priority" Header Field",
draft-ietf-sipcore-priority-00 (work in
progress), December 2012.
[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.
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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.
[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
Schulzrinne, et al. Expires June 21, 2013 [Page 20]