Network Working Group D. Wing
Internet-Draft Cisco
Intended status: Informational S. Fries
Expires: November 23, 2007 Siemens AG
H. Tschofenig
Nokia Siemens Networks
May 22, 2007
Media Security Requirements
draft-wing-media-security-requirements-03.txt
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Copyright (C) The IETF Trust (2007).
Abstract
A number of proposals have been published to address the need of
securing media traffic. Different assumptions, requirements, and
usage environments justify every one of them. This document aims to
summarize the discussed media security requirements in order progress
the work on identifying a small subset applicable to a large range of
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deployment environments.
This document is discussed on the RTPSEC mailing list,
http://www.imc.org/ietf-rtpsec.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Discussion of Call Scenarios . . . . . . . . . . . . . . . . . 3
3.1. Clipping Media Before Signaling Answer . . . . . . . . . . 4
3.2. Retargeting and Forking . . . . . . . . . . . . . . . . . 4
3.3. Shared Key Conferencing . . . . . . . . . . . . . . . . . 7
4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 8
5. Requirements Classification . . . . . . . . . . . . . . . . . 12
6. Security Considerations . . . . . . . . . . . . . . . . . . . 14
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 14
9.1. Normative References . . . . . . . . . . . . . . . . . . . 14
9.2. Informative References . . . . . . . . . . . . . . . . . . 15
Appendix A. Out-of-Scope . . . . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16
Intellectual Property and Copyright Statements . . . . . . . . . . 18
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1. Introduction
The work on media security started a long time ago where the
capability of the Session Initiation Protocol (SIP) was still at its
infancy. With the increased SIP deployment and the availability of
new SIP extensions and related protocols the need for end-to-end
security was re-evaluated. The procedure of re-evaluating prior
protocol work and design decisions is not an uncommon strategy and,
to some extend, considered necessary protocol work to ensure that the
developed protocols indeed meet the previously envisioned needs for
the users in the Internet.
This document aims to summarize the discussed media security
requirements, i.e., requirements for mechanisms that negotiate keys
for SRTP. Once the list of requirements and architectural aspects
have been investigated, the work on the protocol proposals can be
progressed by identifying a small number of soltuions and to complete
the protocol work.
This document is organized as follows. Section 2 introduces
terminology, Section 3 provides an overview about possible call
scenarios, Section 4 lists requirements for media security, Section 5
will provide a clustering of requirements to certain deployment
environments to adress the problem that there might not be a single
solution with universal applicability and Appendix A provides out-of-
scope items and aspects for further discussion. The document
concludes with a security considerations Section 6, IANA
considerations Section 7 and an acknowledgement section in Section 8.
2. Terminology
The keywords "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].
3. Discussion of Call Scenarios
The following subsections describe call scenarios, which have been
discussed elaborately. These call scenarios pose the most challenge
to the key management for media data in cooperation with SIP
signaling. The scenarios have already been described as part of the
key management evaluation draft [I-D.wing-rtpsec-keying-eval], and
are stated here to give a better insight in these discussion.
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3.1. Clipping Media Before Signaling Answer
Per the SDP Offer/Answer Model [RFC3264],
"Once the offerer has sent the offer, it MUST be prepared to
receive media for any recvonly streams described by that offer.
It MUST be prepared to send and receive media for any sendrecv
streams in the offer, and send media for any sendonly streams in
the offer (of course, it cannot actually send until the peer
provides an answer with the needed address and port information)."
To meet this requirement with SRTP, the offerer needs to know the
SRTP key for arriving media. If encrypted SRTP media arrives before
the associated SRTP key, the offerer cannot play the media -- causing
clipping.
For key exchange mechanisms that send the answerer's key in SDP, a
SIP provisional response [RFC3261], such as 183 (session progress),
is useful. However, the 183 messages are not reliable unless both
the calling and called end point support PRACK [RFC3262], use TCP
across all SIP proxies, implement Security Preconditions
[I-D.ietf-mmusic-securityprecondition], or the both ends implement
ICE [I-D.ietf-mmusic-ice] and the answerer implements the reliable
provisional response mechanism described in ICE. Unfortunately,
there is not wide deployment of any of these techniques and there is
industry reluctance to set requirements regarding these techniques to
avoid the problem described in this section.
Note that the receipt of an SDP answer is not always sufficient to
allow media to be played to the offerer. Sometimes, the offerer must
send media in order to open up firewall holes or NAT bindings before
media can be received. In this case a solution that makes the key
available before the SDP answer arrives will not help.
Requirements are created due to early media, in the sense of pre-
offer/answer media, as described in [I-D.barnes-sip-em-ps-req-sol].
Fixes to early media might make the requirements to become obsolete.
3.2. Retargeting and Forking
In SIP, a request sent to a specific AOR but delivered to a different
AOR is called a "retarget". A typical scenario is a "call
forwarding" feature. In Figure 1 Alice sends an Invite in step 1
that is sent to Bob in step 2. Bob responds with a redirect (SIP
response code 3xx) pointing to Carol in step 3. This redirect
typically does not propagate back to Alice but only goes to a proxy
(i.e., the retargeting proxy) that sends the original Invite to Carol
in step 4.
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+-----+
|Alice|
+--+--+
|
| Invite (1)
V
+----+----+
| proxy |
++-+-----++
| ^ |
Invite (2) | | | Invite (4)
& redirect (3) | | |
V | V
++-++ ++----+
|Bob| |Carol|
+---+ +-----+
Figure 1: Retargeting
The mechanism used by SIP for identifying the calling party is SIP
Identity [RFC3261]. However, due to SIP retargeting issues
[I-D.peterson-sipping-retarget], SIP Identity can only identify the
calling party (that is, the party that initiated the SIP request).
Some key exchange mechanisms predate SIP Identity and include their
own identity mechanism. However, those built-in identity mechanism
suffer from the same SIP retargeting problem described in the above
draft. Going forward, it is anticipated that Connected Identity
[I-D.ietf-sip-connected-identity] may allow identifying the called
party. This is also described as the 'retargeting identity' problem.
In SIP, 'forking' is the delivery of a request to multiple locations.
This happens when a single AOR is registered more than once. An
example of forking is when a user has a desk phone, PC client, and
mobile handset all registered with the same AOR.
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+-----+
|Alice|
+--+--+
|
| Invite
V
+-----+-----+
| proxy |
++---------++
| |
Invite | | Invite
V V
+--+--+ +--+--+
|Bob-1| |Bob-2|
+-----+ +-----+
Figure 2: Forking
With forking, both Bob-1 and Bob-2 might send back SDP answers in SIP
responses. Alice will see those intermediate (18x) and final (200)
responses. It is useful for Alice to be able to associate the SIP
response with the incoming media stream. Although this association
can be done with ICE [I-D.ietf-mmusic-ice], and ICE is useful to make
this association with RTP, it is not desirable to require ICE to
accomplish this association.
Forking and retargeting are often used together. For example, a boss
and secretary might have both phones ring and rollover to voice mail
if neither phone is answered.
To maintain security of the media traffic, only the end point that
answers the call should know the SRTP keys for the session. This is
only an issue with public key encryption and not with DH-based
approaches. For key exchange mechanisms that do not provide secure
forking or secure retargeting, one workaround is to re-key
immediately after forking or retargeting (that is, perform a re-
Invite). However, because the originator may not be aware that the
call forked this mechanism requires rekeying immediately after every
session is established. This doubles the Invite messages processed
by the network.
Retargeting securely introduces a more significant problem. With
retargeting, the actual recipient of the request is not the original
recipient. This means that if the offerer encrypted material (such
as the session key or the SDP) using the original recipient's public
key, the actual recipient will not be able to decrypt that material
because the recipient won't have the original recipient's private
key. In some cases, this is the intended behavior, i.e., you wanted
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to establish a secure connection with a specific individual. In
other cases, it is not intended behavior (you want all voice media to
be encrypted, regardless of who answers).
For some forms of key management the calling party needs to know in
advance the certificate or shared secret of the called party, and
retargeting can interfere with this.
Further compounding this problem is a particularity of SIP that when
forking is used, there is always only one final error response
delivered to the sender of the request: the forking proxy is
responsible for choosing which final response to choose in the event
where forking results in multiple final error responses being
received by the forking proxy. This means that if a request is
rejected, say with information that the keying information was
rejected and providing the far end's credentials, it is very possible
that the rejection will never reach the sender. This problem, called
the Heterogeneous Error Response Forking Problem (HERFP)
[I-D.mahy-sipping-herfp-fix], is difficult to solve in SIP.
3.3. Shared Key Conferencing
For efficient scaling, large audio and video conference bridges
operate most efficiently by encrypting the current speaker once and
distributing that stream to the conference attendees. Typically,
inactive participants receive the same streams -- they hear (or see)
the active speaker(s), and the active speakers receive distinct
streams that don't include themselves. In order to maintain
confidentiality of such conferences where listeners share a common
key, all listeners must rekeyed when a listener joins or leaves a
conference.
An important use case for mixers/translators is a conference bridge:
+----+
A --- 1 --->| |
<-- 2 ----| M |
| I |
B --- 3 --->| X |
<-- 4 ----| E |
| R |
C --- 5 --->| |
<-- 6 ----| |
+----+
Figure 3: Centralized Keying
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In the figure above, 1, 3, and 5 are RTP media contributions from
Alice, Bob, and Carol, and 2, 4, and 6 are the RTP flows to those
devices carrying the 'mixed' media.
Several scenarios are possible:
a. Multiple inbound sessions: 1, 3, and 5 are distinct RTP sessions,
b. Multiple outbound sessions: 2, 4, and 6 are distinct RTP
sessions,
c. Single inbound session: 1, 3, and 5 are just different sources
within the same RTP session,
d. Single outbound session: 2, 4, and 6 are different flows of the
same (multi-unicast) RTP session
If there are multiple inbound sessions and multiple outbound sessions
(scenarios a and b), then every keying mechanism behaves as if the
mixer were an end point and can set up a point-to-point secure
session between the participant and the mixer. This is the simplest
situation, but is computationally wasteful, since SRTP processing has
to be done independently for each participant. The use of multiple
inbound sessions (scenario a) doesn't waste computational resources,
though it does consume additional cryptographic context on the mixer
for each participant and has the advantage of non-repudiation of the
originator of the incoming stream.
To support a single outbound session (scenario d), the mixer has to
dictate its encryption key to the participants. Some keying
mechanisms allow the transmitter to determine its own key, and others
allow the offerer to determine the key for the offerer and answerer.
Depending on how the call is established, the offerer might be a
participant (such as a participant dialing into a conference bridge)
or the offerer might be the mixer (such as a conference bridge
calling a participant). The use of offerless Invites may help some
keying mechanisms reverse the role of offerer/answerer. A
difficulty, however, is knowing a priori if the role should be
reversed for a particular call.
4. Requirements
R1: Negotiation of SRTP keys MUST NOT cause the call setup to fail
in forked and retargeted calls where all end points are
willing to use SRTP, unless the execution of the
authentication and key exchange protocol leads to a failure
(e.g., an unsuccessful authentication attempt).
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R2: Even when some end points of a forked or retargeted call are
incapable of using SRTP, the key management protocol MUST
allow the establishment of SRTP associations with SRTP-capable
endpoints and RTP associations with non-SRTP-capable
endpoints.
R3: Forked end points SHOULD NOT know the SRTP key of any call
established with another forked end point. [Editor's Note:
'SHOULD NOT' might be turned into a 'MUST NOT']
R4: A solution MAY support the ability to utilize an initially
established security context that was established as part of
the first call setup with a remote end point.
Specialized devices may need to avoid public key operations or
Diffie-Hellman operations as much as possible because of the
computational cost or because of the additional call setup
delay. For example, it can take a second or two to perform a
Diffie-Hellman operation in certain devices. Examples of
these specialized devices would include some handsets,
intelligent SIMs, and PSTN gateways. For the typical case
because a phone call has not yet been established, ancillary
processing cycles can be utilized to perform the PK or DH
operation; for example, in a PSTN gateway the DSP, which is
not yet involved with typical DSP operations, could be used to
perform the calculation, so as to avoid having the central
host processor perform the calculation. Some devices, such as
handsets, and intelligent SIMs do not have such ancillary
processing capability.
R5: A solution SHOULD avoid clipping media before SDP answer
without requiring PRACK [RFC3262].
R6: A solution MUST provide protection against passive attacks on
the media path and MUST protect against passive attacks of a
SIP proxy that is legitimately routing SIP messages.
R7: A solution MUST be able to support Perfect Forward Secrecy.
R8: A solution MUST support negotiation of the key exchange
algorithm without incurring per-algorithm computational
expense.
R9: A solution MUST support multiple SRTP cipher suites without
additional computational expense.
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R10: A solution that utilizes expensive cryptographic computations
SHOULD offer the ability to resume previous sessions in an
efficient way.
For example, if using a Diffie-Hellman keying technique with
security preconditions that forks to 20 end points, the call
initiator would get 20 provisional responses containing 20
signed Diffie-Hellman key pairs. Calculating 20 DH secrets
and validating signatures can be a difficult task depending on
the device capabilities. Hence, in the case of forking, it is
not desirable to perform a DH or PK operation with every
party, but rather only with the party that answers the call
(and incur some media clipping).
R11: A solution MUST NOT require 3rd parties to sign certificates.
This requirement points to the fact that a global PKI cannot
be assumed and opportunistic security approaches should be
considered in the solution.
R12: A solution SHOULD use algorithms that allow FIPS 140-2
[FIPS-140-2] certification.
Note that the United States Government can only purchase and
use crypto implementations that have been validated by the
FIPS-140 [FIPS-140-2] process:
"The FIPS-140 standard is applicable to all Federal agencies
that use cryptographic-based security systems to protect
sensitive information in computer and telecommunication
systems, including voice systems. The adoption and use of
this standard is available to private and commercial
organizations."[cryptval]
Some commercial organizations, such as banks and defense
contractors, also require or prefer equipment which has
validated by the FIPS-140 process.
R13: A solution for authentication and key exchange SHOULD be able
to associate the signaling exchange with the media traffic.
R14: A solution SHOULD allow to start with RTP and then upgrade to
SRTP.
R15: A solution SHOULD not introduce new denial of service
vulnerabilities.
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R16: A solution SHOULD require the adversary to have access to the
data as well as the signaling path for a successful attack to
be launched. An adversary that is located only along the data
or the signaling path MUST NOT be able to successfully mount
an attack.
R17: A solution SHOULD support the possibility to protect non-RTP-
based data traffic.
R18: A solution MUST provide crypto-agility.
R19: A solution MUST protect cipher suite negotiation against
downgrading attacks.
R20: A solution MUST allow a SIP User Agent to negotiate media
security parameters for each individual session.
R21: A solution SHOULD allow establishing SRTP keying between
different call signaling protocols (e.g., between Jabber, SIP,
H.323, MGCP)
R22: A solution SHOULD support recording of decrypted media.
Media recording may be realized by an intermediate nodes. An
example for those intermediate nodes are devices, which could
be used in banking applications or for quality monitoring in
call centers. Here, it must be ensured, that the media
security is ensured by the intermediate nodes on the
connections to the involved endpoints as originally
negotiated. The endpoints need to be informed that there is
an intermediate device and need to cooperate. A solution
approach for this is described in [I-D.wing-sipping-srtp-key].
R23: A solution SHOULD NOT allow end users to determine whether
their end-to-end interaction is subject to lawful
interception.
R24: A solution MUST work when there are intermediate nodes,
terminating or processing media, between the end points.
[Note: this requirement needs more detail.]
R25: A solution MUST consider termination of media security in a
PSTN gateway.
A typical case of using media security is the one where two
entities are having a VoIP conversation over IP capable
networks. However, there are cases where the other end of the
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communication is not connected to an IP capable network. In
this kind of setting, there needs to be some kind of gateway
at the edge of the IP network which converts the VoIP
conversation to format understood by the other network. An
example of such gateway is a PSTN gateway sitting at the edge
of IP and PSTN networks.
If media security (e.g., SRTP protection) is employed in this
kind of gateway-setting, then media security and the related
key management needs to be terminated at the gateway. The
other network (e.g., PSTN) may have its own measures to
protect the communication, but this means that from media
security point of view the media security is not employed end-
to-end between the communicating entities.
Therefore, media security solutions should cover the cases
where media security is not employed end-to-end but is
terminated in a gateway.
R26: If two parties share an authentication infrastructure that has
3rd parties signing certificates, they MUST be able to use it,
should the endpoints so desire.
[Note: in earlier versions of this document, this
requirement was part of R11.]
R27: If SRTP keying is performed over the media path, the keying
packets MUST NOT pass the RTP validity check defined in
Appendix A.1 of [RFC3550].
5. Requirements Classification
An adversary might be located along
1. the media path,
2. the signaling path,
3. the media and the signaling path.
An attacker that can solely be located along the signaling path, and
does not have access to media, is not considered (ref item 2).
Furthermore, it is reasonable to consider the capabilities of the
adversary. We also have different types of adversaries, namely
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a. active adversary
b. passive adversary
Note that the adversary model for (a) and (b) also assumes the
attacker being able to control SIP signaling entities.
With respect to item (a) an adversary may need to be active with
regard to the key exchange relevant information traveling along the
data or the signaling path.
Some of the deployment variants of the media security key management
proposals under considerations do not provide protection against man-
in-the-middle adversaries under certain conditions, for example when
SIP signaling entities are compromised, when a global PKI is missing
or pre-shared secrets are not exchanged between the end points prior
to the protocol exchange.
Based on the above-mentioned considerations the following
classifications can be made:
Class I:
Passive attack on the signaling and the data path sufficient to
reveal the content of the media traffic.
Class II:
Active attack on the signaling path and passive attack on the data
path to reveal the content of the media traffic.
Class III:
Active attack on the signaling and the data path necessary to
reveal the content of the media traffic.
Class IV:
Active attack is required and will be detected by the end points
when adversary tampers with the messages.
For example, SDES falls into Class I since the adversary needs to
learn the SDES key by progressing a signaling message at a SIP proxy
(assuming that the adversary is in control of the SIP proxy).
Subsequent media traffic can be decrypted with the help of the
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learned key.
As another example, DTLS-RTP falls into Class III when DTLS is used a
public key based ciphersuite with self-signed certificates and
without SIP Identity. An adversary would have to modify the
fingerprint that is sent along the signaling path and subsequently to
modify the certificates carried in the DTLS handshake that travel
along the media path.
An attack is not successful when SIP Identity is used, the adversary
is not between the SIP UA and its Authentication Service (or at the
Authentication Service), both end points are able to verify the
digital signature (of the SIP Identity) and are able to validate the
corresponding certificates.
6. Security Considerations
This document lists requirements for securing media traffic. As
such, it addresses security throughout the document.
7. IANA Considerations
This document does not require actions by IANA.
8. Acknowledgements
The authors would like to thank the active participants of the RTPSEC
BoF and on the RTPSEC mailing list. The authors would furthermore
like to thank Wolfgang Buecker, Guenther Horn, Peter Howard, Hans-
Heinrich Grusdt, Srinath Thiruvengadam, Martin Euchner, Eric
Rescorla, Matt Lepinski, Dan York, Werner Dittmann, Richard Barnes,
Vesa Lehtovirta, Colin Perkins, and Christer Holmberg for their
feedback to this document. We would like to particularly thank
Francois Audet for his work on the evaluation of various media
security key exchange proposals.
9. References
9.1. Normative References
[FIPS-140-2]
NIST, "Security Requirements for Cryptographic Modules",
June 2005, <http://csrc.nist.gov/publications/fips/
fips140-2/fips1402.pdf>.
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[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.
[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of
Provisional Responses in Session Initiation Protocol
(SIP)", RFC 3262, June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[cryptval]
NIST, "Cryptographic Module Validation Program",
December 2006,
<http://csrc.nist.gov/cryptval/140-2APP.htm>.
9.2. Informative References
[I-D.barnes-sip-em-ps-req-sol]
Barnes, R. and M. Lepinski, "Early Media in SIP: Problem
Statement, Requirements, and Analysis of Solutions",
draft-barnes-sip-em-ps-req-sol-00 (work in progress),
February 2007.
[I-D.ietf-mmusic-ice]
Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Methodology for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols",
draft-ietf-mmusic-ice-15 (work in progress), March 2007.
[I-D.ietf-mmusic-securityprecondition]
Andreasen, F. and D. Wing, "Security Preconditions for
Session Description Protocol (SDP) Media Streams",
draft-ietf-mmusic-securityprecondition-03 (work in
progress), October 2006.
[I-D.ietf-sip-connected-identity]
Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", draft-ietf-sip-connected-identity-05
(work in progress), February 2007.
[I-D.mahy-sipping-herfp-fix]
Mahy, R., "A Solution to the Heterogeneous Error Response
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Forking Problem (HERFP) in the Session Initiation
Protocol (SIP)", draft-mahy-sipping-herfp-fix-01 (work in
progress), March 2006.
[I-D.peterson-sipping-retarget]
Peterson, J., "Retargeting and Security in SIP: A
Framework and Requirements",
draft-peterson-sipping-retarget-00 (work in progress),
February 2005.
[I-D.wing-rtpsec-keying-eval]
Audet, F. and D. Wing, "Evaluation of SRTP Keying with
SIP", draft-wing-rtpsec-keying-eval-02 (work in progress),
February 2007.
[I-D.wing-sipping-srtp-key]
Wing, D., "Disclosing Secure RTP (SRTP) Session Keys with
a SIP Event Package", draft-wing-sipping-srtp-key-00 (work
in progress), February 2007.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
Appendix A. Out-of-Scope
Discussions concluded that key management for shared-key encryption
of conferencing is outside the scope of this document. As the
priority is point-to-point unicast SRTP session keying, resolving
shared-key SRTP session keying is deferred to later and left as an
item for future investigations.
Authors' Addresses
Dan Wing
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Email: dwing@cisco.com
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Steffen Fries
Siemens AG
Otto-Hahn-Ring 6
Munich, Bavaria 81739
Germany
Email: steffen.fries@siemens.com
Hannes Tschofenig
Nokia Siemens Networks
Otto-Hahn-Ring 6
Munich, Bavaria 81739
Germany
Email: Hannes.Tschofenig@nsn.com
URI: http://www.tschofenig.com
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