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

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

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   This Internet-Draft will expire on October 22, 2017.

Copyright Notice

   Copyright (c) 2017 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   described in the Simplified BSD License.



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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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