Public Safety Answering Point (PSAP) Callback
draft-ietf-ecrit-psap-callback-03
The information below is for an old version of the document.
| Document | Type | Active Internet-Draft (ecrit WG) | |
|---|---|---|---|
| Authors | Christer Holmberg , Hannes Tschofenig , Henning Schulzrinne , Milan Patel | ||
| Last updated | 2011-10-27 (Latest revision 2010-11-08) | ||
| Replaces | draft-schulzrinne-ecrit-psap-callback | ||
| Stream | Internet Engineering Task Force (IETF) | ||
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draft-ietf-ecrit-psap-callback-03
ECRIT H. Schulzrinne
Internet-Draft Columbia University
Intended status: Standards Track H. Tschofenig
Expires: April 29, 2012 Nokia Siemens Networks
C. Holmberg
Ericsson
M. Patel
InterDigital Communications
October 27, 2011
Public Safety Answering Point (PSAP) Callback
draft-ietf-ecrit-psap-callback-03.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 offers capabilities to allow
callbask to bypass authorization policies to reach the caller without
unnecessary delays. However, the mechanism specified prior to this
document supports only limited scenarios. This document discusses
some shortcomings, presents additional scenarios where better-than-
normal call treatment behavior would be desirable, and specifies a
protocol solution.
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-
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
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time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on April 29, 2012.
Copyright Notice
Copyright (c) 2011 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.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
3. Callback Scenarios . . . . . . . . . . . . . . . . . . . . . . 6
3.1. Routing Asymmetry . . . . . . . . . . . . . . . . . . . . 6
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. Specification . . . . . . . . . . . . . . . . . . . . . . . . 12
5. Security Considerations . . . . . . . . . . . . . . . . . . . 13
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 16
8.1. Normative References . . . . . . . . . . . . . . . . . . . 16
8.2. Informative References . . . . . . . . . . . . . . . . . . 17
Appendix A. Alternative Solutions Considered . . . . . . . . . . 19
A.1. Identity-based Authorization . . . . . . . . . . . . . . . 19
A.2. Trait-based Authorization . . . . . . . . . . . . . . . . 20
A.3. Call Marking . . . . . . . . . . . . . . . . . . . . . . . 21
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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
[I-D.ietf-ecrit-framework] 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 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. [RFC5627] specifies how to obtain and use GRUUs.
Regulatory requirements demand that the emergency call itself
provides enough information to allow the call-taker to initiate a
call back to the emergency caller in case the call dropped or to
interact with the emergency caller in case of further questions. 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 whitelis
or may be forwarded to an answering machine as SIP entities (SIP
proxies as well as the SIP UA itself) cannot differentiate the
callback from any other SIP call establishing attempt from the SIP
signaling message.
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While there are no regulatory requirements at the time of writing of
this specification there is the believe that PSAP callbacks have to
be treated in such a way that they reach the emergency caller. For
this purpose guidance for PSAP callback handling has been provided in
Section 13 of [I-D.ietf-ecrit-framework]:
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. Unfortunately, it does not work in all SIP deployment
scenarios. In Section 3 we describe scenarios where the currently
standardized approach is insufficient. In Section 4 a solution is
described.
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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].
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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 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 intermedidate 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. Specification
[Editor's Note: The solution approach described in
[I-D.holmberg-emergency-callback-id] will be discussed at the IETF#82
ECRIT meeting and at the ECRIT mailing list and will be incorporated
here if agreed by the working group.]
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5. Security Considerations
[Editor's Note: Instead of an abstract security description text will
be provided with the solution description.]
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6. IANA Considerations
[Editor's Note: IANA consideration text will be added once an
agreement on the solution has been reached.
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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.
Finally, we would like to thank Cullen Jennings for his discussion
input. He was the first to propose a "token-based" solution.
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8. References
8.1. Normative References
[RFC2119] Bradner, S., "Key words for use
in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119,
March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H.,
Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R.,
Handley, M., and E. Schooler,
"SIP: Session Initiation
Protocol", RFC 3261, June 2002.
[RFC3325] Jennings, C., Peterson, J., and
M. Watson, "Private Extensions
to the Session Initiation
Protocol (SIP) for Asserted
Identity within Trusted
Networks", RFC 3325,
November 2002.
[RFC3966] Schulzrinne, H., "The tel URI
for Telephone Numbers",
RFC 3966, December 2004.
[RFC3969] Camarillo, G., "The Internet
Assigned Number Authority
(IANA) Uniform Resource
Identifier (URI) Parameter
Registry for the Session
Initiation Protocol (SIP)",
BCP 99, RFC 3969,
December 2004.
[RFC4474] Peterson, J. and C. Jennings,
"Enhancements for Authenticated
Identity Management in the
Session Initiation Protocol
(SIP)", RFC 4474, August 2006.
[RFC5341] Jennings, C. and V. Gurbani,
"The Internet Assigned Number
Authority (IANA) tel Uniform
Resource Identifier (URI)
Parameter Registry",
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September 2008.
[RFC5627] Rosenberg, J., "Obtaining and
Using Globally Routable User
Agent URIs (GRUUs) in the
Session Initiation Protocol
(SIP)", RFC 5627, October 2009.
8.2. Informative References
[I-D.holmberg-emergency-callback-id] Holmberg, C., "Session
Initiation Protocol (SIP)
emergency call back
identification", draft-
holmberg-emergency-callback-id-
00 (work in progress),
October 2011.
[I-D.ietf-ecrit-framework] Rosen, B., Schulzrinne, H.,
Polk, J., and A. Newton,
"Framework for Emergency
Calling using Internet
Multimedia",
draft-ietf-ecrit-framework-13
(work in progress),
September 2011.
[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.
[I-D.ietf-sip-saml] Tschofenig, H., Hodges, J.,
Peterson, J., Polk, J., and D.
Sicker, "SIP SAML Profile and
Binding",
draft-ietf-sip-saml-08 (work in
progress), October 2010.
[RFC4484] Peterson, J., Polk, J., Sicker,
D., and H. Tschofenig, "Trait-
Based Authorization
Requirements for the Session
Initiation Protocol (SIP)",
RFC 4484, August 2006.
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[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.
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Appendix A. Alternative Solutions Considered
In an attempt to describe the problem and to explore solution
approaches the working group had also investigated alternative
approaches. We document them here for completeness. The solutions
fall into three categories: (1) Identity-based authorization, (2)
Trait-based authorization, and (3) Call Marking. Even though these
solutions are not mutually exclusive we describe them in separate
sub-sections.
Beyond the disadvantages listed in each solution category none of
them provides the emergency caller with the ability to restrict
preferential PSAP callback handling to those cases where an earlier
emergency call was initiated.
A.1. Identity-based Authorization
In Figure 6 an interaction is presented that allows a SIP entity to
make a policy decision whether to bypass installed authorization
policies and thereby providing preferential treatment. To make this
decision the sender's identity is compared with a whitelist of valid
PSAPs. The identity assurances 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.
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+----------+
| List of |+
| valid ||
| PSAP ids ||
+----------+|
+----------+
*
* whitelist
*
V
Incoming +----------+ Normal
SIP Msg | SIP |+ Treatment
-------------->| Entity ||=============>
+ Identity | ||(if not in whitelist)
+----------+|
+----------+
||
||
|| Preferential
|| Treatment
++=============>
(in whitelist)
Figure 6: Identity-based Authorization
This approach was not chosen because the establishment of a whitelist
containing PSAP identities is operationally complex and does not
easily scale world wide. Only when there is a local relationship
between the VSP/ASP and the PSAP then populating the whitelist is far
simpler. This would, however, constrain the applicability of the
mechanism considerably.
A.2. Trait-based Authorization
An alternative approach to an identity based authorization model is
outlined in Figure 7. In fact, RFC 4484 [RFC4484] illustrates a
related emergency service use case.
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+----------+
| List of |+
| trust ||
| anchor ||
+----------+|
+----------+
*
*
*
V
Incoming +----------+ Normal
SIP Msg | SIP |+ Treatment
-------------->| Entity ||=============>
+ trait | ||(no indication
+----------+| of PSAP)
+----------+
||
||
|| Preferential
|| Treatment
++=============>
(indicated as
PSAP)
Figure 7: Trait-based Authorization
In a trait-based authorization scenario an incoming SIP message
contains a form of trait, i.e. some form of assertion. The assertion
contains an indication that the sending party has the role of a PSAP
(or similar emergency services entity). The assertion is either
cryptographically protected to enable end-to-end verification or an
chain of trust security model has to be assumed. In Figure 7 we
assume an end-to-end security model where trust anchors are
provisioned to ensure the ability for a SIP entity to verify the
received assertion.
This solution was not chosen because trait-based authorization never
got deployed in SIP. Furthermore, in order to ensure that the
assertions are properly protected it is necessary to digitally sign,
which requires some form of public key infrastructure for usage with
emergency services. Finally, there need to be some policies in place
that define which entities are allowed to obtain various roles.
These policies and procedures do not exist today.
A.3. Call Marking
Call marking allows the PSAP to place a non-cryptographic label on
outgoing calls that gives, when received by a SIP entity,
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preferential treatment for these callbacks.
When used in isolation this mechanism introduces considerable denial
of service attacks due to the ability to bypass any authorization
policies and could be utilized to distribute unwanted traffic.
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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 April 29, 2012 [Page 23]