Internet DRAFT - draft-thomson-mmusic-sdp-uks
draft-thomson-mmusic-sdp-uks
Network Working Group M. Thomson
Internet-Draft E. Rescorla
Intended status: Informational Mozilla
Expires: October 22, 2017 April 20, 2017
Unknown Key Share Attacks on uses of Transport Layer Security with the
Session Description Protocol (SDP)
draft-thomson-mmusic-sdp-uks-00
Abstract
Unknown key-share attacks on the use of Datagram Transport Layer
Security for the Secure Real-Time Transport Protocol (DTLS-SRTP) and
its use with Web Real-Time Communications (WebRTC) identity
assertions are described. Simple mitigation techniques are defined.
Status of This Memo
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This Internet-Draft will expire on October 22, 2017.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Unknown Key-Share Attack . . . . . . . . . . . . . . . . . . 3
2.1. Attack Overview . . . . . . . . . . . . . . . . . . . . . 3
2.2. Limits on Attack Feasibility . . . . . . . . . . . . . . 4
2.3. Example . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.4. Interactions with Key Continuity . . . . . . . . . . . . 6
3. Adding a Session Identifier . . . . . . . . . . . . . . . . . 6
3.1. The sdp_tls_id TLS Extension . . . . . . . . . . . . . . 7
4. WebRTC Identity Binding . . . . . . . . . . . . . . . . . . . 8
4.1. The webrtc_id_hash TLS Extension . . . . . . . . . . . . 8
5. Session Concatenation . . . . . . . . . . . . . . . . . . . . 9
6. Security Considerations . . . . . . . . . . . . . . . . . . . 10
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 11
8.1. Normative References . . . . . . . . . . . . . . . . . . 11
8.2. Informative References . . . . . . . . . . . . . . . . . 12
Appendix A. Acknowledgements . . . . . . . . . . . . . . . . . . 13
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 13
1. Introduction
The use of Transport Layer Security (TLS) [RFC5246] with the Session
Description Protocol (SDP) [RFC4566] is defined in [RFC4572].
Further use with Datagram Transport Layer Security (DTLS) [RFC6347]
and the Secure Real-time Transport Protocol (SRTP) [RFC3711] is
defined as DTLS-SRTP [RFC5763].
In these specifications, key agreement is performed using TLS or
DTLS, with authentication being tied back to the session description
(or SDP) through the use of certificate fingerprints. Communication
peers check that a hash, or fingerprint, provided in the SDP matches
the certificate that is used in the TLS or DTLS handshake. This is
defined in [RFC4572].
The design in RFC 4572 relies on the integrity of the signaling
channel. Certificate fingerprints are assumed to be provided by the
communicating peers and carried by the signaling channel without
being subject to modification. However, this design is vulnerable to
an unknown key-share (UKS) attack where a misbehaving endpoint is
able to advertise a key that it does not control. This leads to the
creation of sessions where peers are confused about the identify of
the participants.
An extension to TLS is defined that can be used to mitigate this
attack.
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A similar attack is possible with sessions that use WebRTC identity
(see Section 5.6 of [I-D.ietf-rtcweb-security-arch]). This issue and
a mitigation for it is discussed in more detail in Section 4.
2. Unknown Key-Share Attack
In an unknown key-share attack [UKS], a malicious participant in a
protocol claims to control a key that is in reality controlled by
some other actor. This arises when the identity associated with a
key is not properly bound to the key.
In usages of TLS and DTLS that use SDP for negotiation, an endpoint
is able to acquire the certificate fingerprint another entity. By
advertising that fingerprint in place of one of its own, the
malicious endpoint can cause its peer to communicate with a different
peer, even though it believes that it is communicating with the
malicious endpoint.
When the identity of communicating peers is established by higher-
layer signaling constructs, such as those in SIP [RFC4474] or WebRTC
[I-D.ietf-rtcweb-security-arch], this allows an attacker to bind
their own identity to a session with any other entity.
By substituting the the fingerprint of one peer for its own, an
attacker is able to cause a session to be established where one
endpoint has an incorrect value for the identity of its peer.
However, the peer does not suffer any such confusion, resulting in
each peer involved in the session having a different view of the
nature of the session.
This attack applies to any communications established based on the
SDP "fingerprint" attribute [RFC4572].
2.1. Attack Overview
This vulnerability can be used by an attacker to create a session
where there is confusion about the communicating endpoints.
A SIP endpoint or WebRTC endpoint that is configured to reuse a
certificate can be attacked if it is willing to conduct two
concurrent calls, one of which is with an attacker. The attacker can
arrange for the victim to incorrectly believe that is calling the
attacker when it is in fact calling a second party. The second party
correctly believes that it is talking to the victim.
In a related attack, a single call using WebRTC identity can be
attacked so that it produces the same outcome. This attack does not
require a concurrent call.
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2.2. Limits on Attack Feasibility
The use of TLS with SDP depends on the integrity of session
signaling. Assuming signaling integrity limits the capabilities of
an attacker in several ways. In particular:
1. An attacker can only modify the parts of the session signaling
for a session that they are part of, which is limited to their
own offers and answers.
2. No entity will complete communications with a peer unless they
are willing to participate in a session with that peer.
The combination of these two constraints make the spectrum of
possible attacks quite limited. An attacker is only able to switch
its own certificate fingerprint for a valid certificate that is
acceptable to its peer. Attacks therefore rely on joining two
separate sessions into a single session.
The second condition is not necessary with WebRTC identity if the
victim has or is configured with a target peer identity (this is
defined in [WEBRTC]). Furthermore, any identity displayed by a
browser could be different to the identity used by the application,
since the attack affects the browser's understanding of the peer's
identity.
2.3. Example
In this example, two outgoing sessions are created by the same
endpoint. One of those sessions is initiated with the attacker,
another session is created toward another honest endpoint. The
attacker convinces the endpoint that their session has completed, and
that the session with the other endpoint has succeeded.
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Norma Mallory Patsy
(fp=N) ----- (fp=P)
| | |
+---Offer1 (fp=N)--->| |
+-----Offer2 (fp=N)-------------------->|
|<--------------------Answer2 (fp=P)----+
|<--Answer1 (fp=P)---+ |
| | |
|======DTLS1====>(Forward)====DTLS1====>|
|<=====DTLS2=====(Forward)<===DTLS2=====|
|======Media1===>(Forward)====Media1===>|
|<=====Media2====(Forward)<===Media2====|
| | |
|======DTLS2===========>(Drop) |
| | |
In this case, Norma is willing to conduct two concurrent sessions.
The first session is established with Mallory, who falsely uses
Patsy's certificate fingerprint. A second session is initiated
between Norma and Patsy. Signaling for both sessions is permitted to
complete.
Once complete, the session that is ostensibly between Mallory and
Norma is completed by forwarding packets between Norma and Patsy.
This requires that Mallory is able to intercept DTLS and media
packets from Patsy so that they can be forwarded to Norma at the
transport addresses that Norma associates with the first session.
The second session - between Norma and Patsy - is permitted to
continue to the point where Patsy believes that it has succeeded.
This ensures that Patsy believes that she is communicating with
Norma. In the end, Norma believes that she is communicating with
Mallory, when she is actually communicating with Patsy.
Though Patsy needs to believe that the second session is successful,
Mallory has no real interest in seeing that session complete.
Mallory only needs to ensure that Patsy does not abandon the session
prematurely. For this reason, it might be necessary to permit the
answer from Patsy to reach Norma to allow Patsy to receive a call
completion signal, such as a SIP ACK. Once the second session
completes, Mallory causes any DTLS packets sent by Norma to Patsy to
be dropped.
For the attacked session to be sustained beyond the point that Norma
detects errors in the second session, Mallory also needs to block any
signaling that Norma might send to Patsy asking for the call to be
abandoned. Otherwise, Patsy might receive a notice that the call is
failed and thereby abort the call.
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This attack creates an asymmetry in the beliefs about the identity of
peers. However, this attack is only possible if the victim (Norma)
is willing to conduct two sessions concurrently, and if the same
certificate - and therefore SDP "fingerprint" attribute value - is
used in both sessions.
2.4. Interactions with Key Continuity
Systems that use key continuity might be able to detect an unknown
key-share attack if a session with the actual peer (i.e., Patsy in
the example) was established in the past. Whether this is possible
depends on how key continuity is implemented.
Implementations that maintain a single database of identities with an
index on peer keys could discover that the identity saved for the
peer key does not match the claimed identity. Such an implementation
could notice the disparity between the actual keys (Patsy) and the
expected keys (Mallory).
In comparison, implementations that first match based on peer
identity could treat an unknown key-share attack as though their peer
had used a newly-configured device. The apparent addition of a new
device could generate user-visible notices (e.g., "Mallory appears to
have a new device"). However, such an event is not always considered
alarming; some implementations might silently save a new key.
3. Adding a Session Identifier
An attack on DTLS-SRTP is possible because the identity of peers
involved is not established prior to establishing the call.
Endpoints use certificate fingerprints as a proxy for authentication,
but as long as fingerprints are used in multiple calls, they are
vulnerable to attacks of the sort described.
The solution to this problem is to assign a new identifier to
communicating peers. Each endpoint assigns their peer a unique
identifier during call signaling. The peer echoes that identifier in
the TLS handshake, binding that identity into the session. Including
this new identity in the TLS handshake means that it will be covered
by the TLS Finished message, which is necessary to authenticate it
(see [SIGMA]). Validating that peers use the correct identifier then
means that the session is established between the correct two
endpoints.
This solution relies on the unique identifier given to DTLS sessions
using the SDP "tls-id" attribute [I-D.ietf-mmusic-dtls-sdp]. This
field is already required to be unique. Thus, no two offers or
answers from the same client will have the same value.
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A new "sdp_tls_id" extension is added to the TLS or DTLS handshake
for connections that are established as part of the same call or
real-time session. This carries the value of the "tls-id" attribute
and provides integrity protection for its exchange as part of the TLS
or DTLS handshake.
3.1. The sdp_tls_id TLS Extension
The "sdp_tls_id" TLS extension carries the unique identifier that an
endpoint selects. The value includes the "tls-id" attribute from the
SDP that the endpoint generated when negotiating the session.
The "extension_data" for the "sdp_tls_id" extension contains a
SdpTlsId struct, described below using the syntax defined in
[RFC5246]:
struct {
opaque tls_id<20..255>;
} SdpTlsId;
The "tls_id" field of the extension includes the value of the "tls-
id" SDP attribute as defined in [I-D.ietf-mmusic-dtls-sdp] (that is,
the "tls-id-value" ABNF production). The value of the "tls-id"
attribute is encoded using ASCII [RFC0020].
Where RTP and RTCP [RFC3550] are not multiplexed, it is possible that
the two separate DTLS connections carrying RTP and RTCP can be
switched. This is considered benign since these protocols are
usually distinguishable. RTP/RTCP multiplexing is advised to address
this problem.
The "sdp_tls_id" extension is included in a ClientHello and either
ServerHello (for TLS and DTLS versions less than 1.3) or
EncryptedExtensions (for TLS 1.3). In TLS 1.3, the "sdp_tls_id"
extension MUST NOT be included in a ServerHello.
Endpoints MUST check that the "tls_id" parameter in the extension
that they receive includes the "tls-id" attribute value that they
received in their peer's session description. Comparison can be
performed with either the decoded ASCII string or the encoded octets.
An endpoint that receives a "sdp_tls_id" extension that is not
identical to the value that it expects MUST abort the connection with
a fatal "handshake_failure" alert.
An endpoint that is communicating with a peer that does not support
this extension will receive a ClientHello, ServerHello or
EncryptedExtensions that does not include this extension. An
endpoint MAY choose to continue a session without this extension in
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order to interoperate with peers that do not implement this
specification.
In TLS 1.3, the "sdp_tls_id" extension MUST be sent in the
EncryptedExtensions message.
4. WebRTC Identity Binding
The identity assertion used for WebRTC
[I-D.ietf-rtcweb-security-arch] is bound only to the certificate
fingerprint of an endpoint and can therefore be copied by an attacker
along with any SDP "fingerprint" attributes.
The problem is compounded by the fact that an identity provider is
not required to verify that the entity requesting an identity
assertion controls the keys. Nor is it currently able to perform
this validation. This is not an issue because verification is not a
necessary condition for a secure protocol, nor would it be sufficient
as established in [SIGMA].
A simple solution to this problem is suggested by [SIGMA]. The
identity of endpoints is included under a message authentication code
(MAC) during the cryptographic handshake. Endpoints are then
expected to validate that their peer has provided an identity that
matches their expectations.
In TLS, the Finished message provides a MAC over the entire
handshake, so that including the identity in a TLS extension is
sufficient to implement this solution. Rather than include a
complete identity assertion - which could be sizeable - a collision-
resistant hash of the identity assertion is included in a TLS
extension. Peers then need only validate that the extension contains
a hash of the identity assertion they received in signaling in
addition to validating the identity assertion.
Endpoints MAY use the "sdp_tls_id" extension in addition to this so
that two calls between the same parties can't be altered by an
attacker.
4.1. The webrtc_id_hash TLS Extension
The "webrtc_id_hash" TLS extension carries a hash of the identity
assertion that communicating peers have exchanged.
The "extension_data" for the "webrtc_id_hash" extension contains a
WebrtcIdentityHash struct, described below using the syntax defined
in [RFC5246]:
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struct {
opaque assertion_hash<0..32>;
} WebrtcIdentityHash;
A WebRTC identity assertion is provided as a JSON [RFC7159] object
that is encoded into a JSON text. The resulting string is then
encoded using UTF-8 [RFC3629]. The content of the "webrtc_id_hash"
extension are produced by hashing the resulting octets with SHA-256
[FIPS180-2]. This produces the 32 octets of the assertion_hash
parameter, which is the sole contents of the extension.
The SDP "identity" attribute includes the base64 [RFC4648] encoding
of the same octets that were input to the hash. The "webrtc_id_hash"
extension is validated by performing base64 decoding on the value of
the SDP "identity" attribute, hashing the resulting octets using SHA-
256, and comparing the results with the content of the extension.
Identity assertions might be provided by only one peer. An endpoint
that does not produce an identity assertion MUST generate an empty
"webrtc_id_hash" extension in its ClientHello. This allows its peer
to include a hash of its identity assertion. An endpoint without an
identity assertion MUST omit the "webrtc_id_hash" extension from its
ServerHello or EncryptedExtensions message.
A peer that receives a "webrtc_id_hash" extension that is not equal
to the value of the identity assertion from its peer MUST immediately
fail the TLS handshake with an error. This includes cases where the
"identity" attribute is not present in the SDP.
A "webrtc_id_hash" extension that is any length other than 0 or 32 is
invalid and MUST cause the receiving endpoint to generate a fatal
"decode_error" alert.
A peer that receives an identity assertion, but does not receive a
"webrtc_id_hash" extension MAY choose to fail the connection, though
it is expected that implementations that were written prior to the
existence of this document will not support these extensions for some
time.
In TLS 1.3, the "webrtc_id_hash" extension MUST be sent in the
EncryptedExtensions message.
5. Session Concatenation
Use of session identifiers does not prevent an attacker from
establishing two concurrent sessions with different peers and
forwarding signaling from those peers to each other. Concatenating
two signaling sessions creates a situation where both peers believe
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that they are talking to the attacker when they are talking to each
other.
Session concatention is possible at higher layers: an attacker can
establish two independent sessions and simply forward any data it
receives from one into the other. This kind of attack is prevented
by systems that enable peer authentication such as WebRTC identity
[I-D.ietf-rtcweb-security-arch] or SIP identity [RFC4474].
In the absence of any higher-level concept of peer identity, the use
of session identifiers does not prevent session concatenation. The
value to an attacker is limited unless information from the TLS
connection is extracted and used with the signaling. For instance, a
key exporter [RFC5705] might be used to create a shared secret or
unique identifier that is used in a secondary protocol.
If a secondary protocol uses the signaling channel with the
assumption that the signaling and TLS peers are the same then that
protocol is vulnerable to attack. The identity of the peer at the
TLS layer is not guaranteed to be the same as the identity of the
signaling peer.
It is important to note that multiple connections can be created
within the same signaling session. An attacker might concatenate
only part of a session, choosing to terminate some connections (and
optionally forward data) while arranging to have peers interact
directly for other connections. It is even possible to have
different peers interact for each connection. This means that the
actual identity of the peer for one connection might differ from the
peer on another connection.
Information extracted from a TLS connection therefore MUST NOT be
used in a secondary protocol outside of that connection if that
protocol relies on the signaling protocol having the same peers.
Similarly, data from one TLS connection MUST NOT be used in other TLS
connections even if they are established as a result of the same
signaling session.
6. Security Considerations
This entire document contains security considerations.
7. IANA Considerations
This document registers two extensions in the TLS "ExtensionType
Values" registry established in [RFC5246]:
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o The "sdp_tls_id" extension has been assigned a code point of TBD;
it is recommended and is marked as "Encrypted" in TLS 1.3.
o The "webrtc_id_hash" extension has been assigned a code point of
TBD; it is recommended and is marked as "Encrypted" in TLS 1.3.
8. References
8.1. Normative References
[FIPS180-2]
Department of Commerce, National., "NIST FIPS 180-2,
Secure Hash Standard", August 2002.
[I-D.ietf-mmusic-dtls-sdp]
Holmberg, C. and R. Shpount, "Using the SDP Offer/Answer
Mechanism for DTLS", draft-ietf-mmusic-dtls-sdp-23 (work
in progress), April 2017.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-12 (work in progress), June 2016.
[RFC0020] Cerf, V., "ASCII format for network interchange", STD 80,
RFC 20, DOI 10.17487/RFC0020, October 1969,
<http://www.rfc-editor.org/info/rfc20>.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
2003, <http://www.rfc-editor.org/info/rfc3629>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572,
DOI 10.17487/RFC4572, July 2006,
<http://www.rfc-editor.org/info/rfc4572>.
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[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246,
DOI 10.17487/RFC5246, August 2008,
<http://www.rfc-editor.org/info/rfc5246>.
[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
2010, <http://www.rfc-editor.org/info/rfc5763>.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
January 2012, <http://www.rfc-editor.org/info/rfc6347>.
8.2. Informative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474,
DOI 10.17487/RFC4474, August 2006,
<http://www.rfc-editor.org/info/rfc4474>.
[RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data
Encodings", RFC 4648, DOI 10.17487/RFC4648, October 2006,
<http://www.rfc-editor.org/info/rfc4648>.
[RFC5705] Rescorla, E., "Keying Material Exporters for Transport
Layer Security (TLS)", RFC 5705, DOI 10.17487/RFC5705,
March 2010, <http://www.rfc-editor.org/info/rfc5705>.
[RFC7159] Bray, T., Ed., "The JavaScript Object Notation (JSON) Data
Interchange Format", RFC 7159, DOI 10.17487/RFC7159, March
2014, <http://www.rfc-editor.org/info/rfc7159>.
[SIGMA] Krawczyk, H., "SIGMA: The 'SIGn-and-MAc'approach to
authenticated Diffie-Hellman and its use in the IKE
protocols", Annual International Cryptology Conference,
Springer, pp. 400-425 , 2003.
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[UKS] Blake-Wilson, S. and A. Menezes, "Unknown Key-Share
Attacks on the Station-to-Station (STS) Protocol", Lecture
Notes in Computer Science 1560, Springer, pp. 154-170 ,
1999.
[WEBRTC] Bergkvist, A., Burnett, D., Narayanan, A., Jennings, C.,
and B. Aboba, "WebRTC 1.0: Real-time Communication Between
Browsers", W3C WD-webrtc-30160531 , May 2016.
Appendix A. Acknowledgements
This problem would not have been discovered if it weren't for
discussions with Sam Scott, Hugo Krawczyk, and Richard Barnes. A
solution similar to the one presented here was first proposed by
Karthik Bhargavan who provided valuable input on this document.
Thyla van der Merwe assisted with a formal model of the solution.
Adam Roach and Paul E. Jones provided useful review and input.
Authors' Addresses
Martin Thomson
Mozilla
Email: martin.thomson@gmail.com
Eric Rescorla
Mozilla
Email: ekr@rftm.com
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