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Versions: 00 01 02

Internet Engineering Task Force                              A. Johnston
Internet-Draft                                                     Avaya
Intended status: Informational                             P. Zimmermann
Expires: January 7, 2016                                       J. Callas
                                                           Silent Circle
                                                                T. Cross
                                                              OfficeTone
                                                               J. Yoakum
                                                                   Avaya
                                                            July 6, 2015


                      Using ZRTP to Secure WebRTC
                     draft-johnston-rtcweb-zrtp-02

Abstract

   WebRTC, Web Real-Time Communications, is a set of protocols and APIs
   used to enable web developers to add real-time communications into
   their web pages and applications with a few lines of JavaScript.
   WebRTC media flows are encrypted and authenticated by SRTP, the
   Secure Real-time Transport Protocol while the key agreement is
   provided by DTLS-SRTP, Datagram Transport Layer Security for Secure
   Real-time Transport Protocol.  However, without some third party
   identity service or certificate authority, WebRTC media flows have no
   protection against a man-in-the-middle (MitM) attack.  ZRTP, Media
   Path Key Agreement for Unicast Secure RTP, RFC 6189, does provide
   protection against MitM attackers using key continuity augmented with
   a Short Authentication String (SAS).  This specification describes
   how ZRTP can be used over the WebRTC data channel to provide MitM
   protection for WebRTC media flows keyed using DTLS-SRTP.  This
   provides users protection against MitM attackers without requiring
   browsers to support ZRTP or users to download a plugin or extension
   to implement ZRTP.

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 January 7, 2016.

Copyright Notice

   Copyright (c) 2015 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  . . . . . . . . . . . . . . . . . . . . . . . .   2
     1.1.  Requirements Language . . . . . . . . . . . . . . . . . .   4
   2.  ZRTP over a WebRTC Data Channel . . . . . . . . . . . . . . .   4
   3.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   5
   4.  Security Considerations . . . . . . . . . . . . . . . . . . .   5
   5.  Implementation Status . . . . . . . . . . . . . . . . . . . .   7
   6.  Appendix A: ZRTP JSON Encoding  . . . . . . . . . . . . . . .   8
   7.  Informative References  . . . . . . . . . . . . . . . . . . .   8
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  10

1.  Introduction

   WebRTC, Web Real-Time Communications, adds real-time, interactive
   voice and video media capabilities to browsers
   [I-D.ietf-rtcweb-overview] without a plugin or download, and allows
   web developers to access this functionality using JavaScript API
   calls [WebRTC-API].  For a complete description of WebRTC protocols
   and APIs see [WebRTC-Book].  In addition, WebRTC supports the
   establishment of a peer-to-peer data channel between browsers
   [I-D.ietf-rtcweb-data-channel].  This document describes how ZRTP,
   Media Path Key Agreement for Unicast Secure RTP, [RFC6189], can be
   used over the WebRTC data channel to secure voice and video sessions
   established using WebRTC.

   The security of voice and video media sessions established using
   WebRTC is described in [I-D.ietf-rtcweb-security].  All media



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   sessions utilize SRTP encryption and authentication, which relies on
   DTLS-SRTP for key management.  DTLS-SRTP can utilize TLS modes
   offering perfect forward secrecy (PFS), but relies on the exchange of
   fingerprints for protection against Man-in-the-Middle (MitM) attacks
   [RFC5763].  A mechanism for utilizing trusted third parties, known as
   Identity Providers, to authenticate the fingerprint is also
   described.  ZRTP always offers perfect forward secrecy, and protects
   against MitM attacks with key continuity, Short Authentication
   Strings (SAS), and optionally and additionally, with long-term
   signing keys or shared secrets.  For subsequent calls between the
   same ZRTP endpoints, a hash of previous keying material is mixed in
   when generating the current keying material.  In addition, the SAS
   can be used to confirm the absence of a MitM attack over the entire
   lifetime of the key continuity (going both backwards and forwards in
   time).  Both parties in the communication must have ZRTP software,
   which performs a DH key agreement and are capable of storing a cache
   of previous shared secrets and rendering the SAS to the users.  The
   human users then have the option to compare the SAS's to see if they
   match to confirm the absence of a MitM attacker.  This could be done
   by verbally reading aloud the string (which can be two words or four
   hex characters), or otherwise exchanging them.  If the SAS values
   match, then there is no MitM attacker.  ZRTP is signaling channel and
   protocol independent, and does not rely on ANY third party services
   for authentication (though it can optionally and additionally
   leverage a public key infrastructure (PKI)).  As such, ZRTP has been
   used with SIP, Jingle, and proprietary signaled VoIP systems.  There
   are a number of open source ZRTP stacks and commercial
   implementations and products.  For the reasons why ZRTP is a good fit
   for WebRTC, see [I-D.johnston-rtcweb-media-privacy].

   ZRTP is not currently built into the browser like DTLS-SRTP.
   However, this doesn't mean that ZRTP cannot be used with WebRTC.
   ZRTP can be implemented in JavaScript and run over the WebRTC data
   channel between the browsers.  The format and message flow can be
   identical to RFC 6189, with the exception that instead of ZRTP
   running on UDP, it runs on top of SCTP/DTLS/UDP.  A small change in
   the policy usage of the ZRTP auxsecret provides MitM protection for
   media sessions established by WebRTC between the browsers.

   This allows the ZRTP SAS to be used to authenticate WebRTC media
   sessions for WebRTC applications that include ZRTP JavaScript.  Also,
   since the ZRTP data channel can be used to authenticate all WebRTC
   Peer Connections between a pair of browsers, a ZRTP WebRTC
   application could be used to authenticate and protect other WebRTC
   sessions that do not even use ZRTP.  For example, users of a
   particular WebRTC service which claims to offer end-to-end media
   privacy could use a ZRTP-enabled WebRTC application in another tab or




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   window to verify that assertion or audit the service and protect
   against MitM attacks.

1.1.  Requirements Language

   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 RFC 2119 [RFC2119].

2.  ZRTP over a WebRTC Data Channel

   In the base ZRTP protocol [RFC6189], ZRTP uses UDP transport,
   multiplexed over the same port as the media session that it is
   keying.  ZRTP over a WebRTC data channel means that ZRTP messages are
   sent over the SCTP/DTLS/UDP protocol stack.  It is RECOMMENDED that
   SCTP reliability be used so that the ZRTP timer and retransmissions
   in Section 6 of [RFC6189] are not needed.  The state machine is
   identical, with the exchange beginning with the Hello and ending with
   the ConfACK.  The ZRTP Hello Hash MAY be exchanged over the WebRTC
   signaling channel.  The ZID MAY be statelessly generated by hashing
   the DTLS-SRTP fingerprint of the browser.  Also, the ZRTP cache of
   previous shared secrets can be stored in a number of ways, including
   indexed database, HTML5 file system, or even as a cookie.

   In order to provide protection against a MitM attack of WebRTC media
   sessions, ZRTP needs to:

   o  Verify that both browsers see the same local and remote
      fingerprint used by DTLS-SRTP.  This is accomplished by always
      including the DTLS-SRTP fingerprints in the ZRTP auxsecret.

   o  Verify that there is no MitM attack against ZRTP.  This is
      accomplished by the various mechanisms ZRTP provides, including
      key continuity and human users comparing the SAS.

   The ZRTP auxsecret is defined in Section 4.3 of [RFC6189].  This
   specification defines the following new policies relating to the
   usage of auxsecret when ZRTP is used to secure DTLS-SRTP media
   sessions.

   The auxsecret MUST be used.  The auxsecret is truncated to the
   negotiated hash length (defined in Section 4.5.1 of [RFC6189]) of:

        auxsecret = hash(initiator's DTLS-SRTP fingerprint ||
                         responder's DTLS-SRTP fingerprint ||
                         original_auxsecret)





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   The original_auxsecret is any auxsecret value that would otherwise
   have been used with ZRTP, or the null string if no such value exists
   as will ordinarily be the case.

   Note that this auxsecret is actually not a secret, since the
   fingerprints are hashes of known public keys used by the browsers.
   This does not affect the security of ZRTP.

   If the auxsecrets of the initiator and responder do not match, this
   MUST be treated as a MitM attack.  This is to protect against the
   case where the DTLS-SRTP session has an MitM attacker but the ZRTP
   session does not.  Note that this can be done as soon as the DHPart1
   and DHPart2 messages have been exchanged and can be done
   automatically without calculating or comparing the SAS.

   Any failure in the ZRTP exchange MUST be treated as a MitM attack.

   Detection of a MitM attack MUST result in the closure of the DTLS-
   SRTP sessions and alerting the browser users.

   If the users successfully compare the SAS strings, it means that
   neither the DTLS nor the ZRTP sessions have MitM attackers.  Any
   media sessions which were established using this same pair of local
   and remote fingerprints also do not have MitM attackers, regardless
   of which browser tab or window they are present in.

   This specification requires DTLS to use a Forward Secrecy (FS) mode.
   If a FS mode is not available, the DTLS connection MUST fail.

3.  IANA Considerations

   This memo includes no request to IANA.

4.  Security Considerations

   For the security analysis of this approach, consider a pair of
   browsers, used by Alice and Bob which have established at a minimum a
   voice media session and a ZRTP data channel.  There are two
   possibilities:

   o  Both the media and data run over the same DTLS connection, or

   o  The media and data run over separate DTLS connections.

   As such, an attacker could choose to attack any combination of these
   connections and the DTLS and/or ZRTP protocols.  However, note that
   since ZRTP runs on top of DTLS, it is not possible to MitM ZRTP
   without first launching a MitM attack on the DTLS connection over



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   which it runs.  In the following analysis, "attacking the media
   channel" means a MitM attack launched against the DTLS session used
   to establish the voice media session, and "attacking the data
   channel" means a MitM attack against ZRTP and the DTLS session over
   which ZRTP runs.

   Given these two possibilities, the attacker could choose to attack:

   o  Both the media and data channel,

   o  Just the media channel,

   o  Just the data channel, or

   o  Neither media or data channel.

   These will be considered in turn.  Note that a MitM attack launched
   against DTLS-SRTP will result in the remote fingerprint as seen by
   each browser to be that of the attacker instead of the other browser.

   If the MitM attacks both the media and the data channel, the SAS as
   computed by each browser will be different, and the users can detect
   this by verbally comparing the SAS.  Additionally, if the users have
   communicated before without a MitM attacker, the presence of the MitM
   will create a break in key continuity and the users will be alerted
   that they should verify the SAS.

   If the MitM attacks just the media channel, after the exchange of
   DHPart1 and DHPart2 messages, the different fingerprints will be
   detected by checking the hashed auxsecret values and discovering that
   they do not match.  The MitM attack is immediately and automatically
   detected.

   If the MitM attacks just the data channel, the SAS as computed by
   each browser will be different as two independent DH exchanges
   occurred.  If the users have spoken before, the MitM will cause a
   break in key continuity.  In any case, the MitM will be definitively
   detected by comparing ZRTP's SAS.  Note that it doesn't make much
   sense for the MitM to attack just the data channel, but this could
   happen.

   If the MitM attacks neither the media nor the data channel, the
   auxsecrets will match, the SAS as computed by each browser will be
   the same, and key continuity will be maintained.  As a result, both
   the ZRTP and media session are free of MitM attackers.

   Note that only in one scenario does this approach rely on the users
   comparing the SAS -- and even there, the users would likely be



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   protected by key continuity even if the SAS were not manually
   checked.  Also, note that all these attacks rely on the attacker
   being able to insert herself in the path as a MitM.  For the scenario
   in which the media channel and data channel use different DTLS
   connections, it could be potentially difficult for the attacker to
   insert herself as a MitM in the data channel as it could take a
   complete different route over the Internet from the media channel.
   For example, the data channel used by ZRTP could be deliberately
   routed over a different IP connection or via a TURN server forcing a
   different path that may not accessible to the attacker.

   In summary, this approach can be thought of as having three distinct
   layers.  The first layer is the DTLS session, which protects against
   passive attacks but has no protection against a MitM attack without a
   third party service.  The next layer is the ZRTP session, which
   allows the fingerprints to be exchanged and compared.  A fingerprint
   mismatch allows a MitM attack on DTLS to be detected.  The third
   layer is ZRTP and its protections against a MitM: short
   authentication strings, key continuity, and optional SAS signing with
   a PKI.  These protections are cumulative -- even over time.  Because
   of key continuity, a single comparison of the SAS guarantees that no
   MitM has attacked past sessions and cannot attack future sessions.
   And even if the SAS is not compared, key continuity ensures that for
   a MitM attacker to remain undetected, she must attack each session
   between the users without exception.

5.  Implementation Status

   Note to RFC Editor: Please remove this entire section prior to
   publication, including the reference to RFC 6982.

   This section records the status of known implementations of the
   protocol defined by this specification at the time of posting of this
   Internet-Draft, and is based on a proposal described in [RFC6982].
   The description of implementations in this section is intended to
   assist the IETF in its decision processes in progressing drafts to
   RFCs.  Please note that the listing of any individual implementation
   here does not imply endorsement by the IETF.  Furthermore, no effort
   has been spent to verify the information presented here that was
   supplied by IETF contributors.  This is not intended as, and must not
   be construed to be, a catalog of available implementations or their
   features.  Readers are advised to note that other implementations may
   exist.

   According to [RFC6982], "this will allow reviewers and working groups
   to assign due consideration to documents that have the benefit of
   running code, which may serve as evidence of valuable experimentation
   and feedback that have made the implemented protocols more mature.



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   It is up to the individual working groups to use this information as
   they see fit".

   An implementation of ZRTP over the data channel was developed for the
   TADHack 2015 Hackathon [TADHACK].  The preliminary implementation of
   the ZRTP JavaScript library, zrtp4js, was developed by Werner
   Dittman, using the Stanford JavaScript Crypto Library [SJCL] while
   the JavaScript application was developed by Dan Burnett, Alan
   Johnston, and Mahak Patel.  A video is available at [TADHACK-ZRTP].
   The next section shows an example of a ZRTP message from that
   implementation.

6.  Appendix A: ZRTP JSON Encoding

   For ZRTP running over the data channel between two browsers, a
   JavaScript Object Notation (JSON) encoding for ZRTP messages will
   simplify the JavaScript parsing and allow the "WebRTC String" PPID to
   be used over the data channel [I-D.ietf-rtcweb-data-channel].  Below
   is an example of an ZRTP DHPart1 message.

   {
           "_data": {
                   "_zrtpId":20570,
                   "_length":0,
                   "_messageType":"DHPart1",
                   "_pv":"{\"x\":[9238189,9730715,15220722,
                           188609,9862612,13314338,
                           6663378,7562888,1865421,
                           11918668,16604863,755888,
                           14892086,11391512,6379092,210972],
                           \"y\":[14957537,2388710,8838444,
                           7786690,12191288,4191233,
                           10790862,1805965,14966182,
                           2627552,14359332,2309769,
                           8590080,6006833,7363525,1740185]}",
                   "_aux":"[1622940278,772233674,93017422,
                           -1125510277,-2095646614,801724212,
                           -663866781,800173874]"
           }
   }

7.  Informative References

   [I-D.ietf-rtcweb-data-channel]
              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", draft-ietf-rtcweb-data-channel-13 (work in
              progress), January 2015.




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   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-14
              (work in progress), June 2015.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-08 (work in progress), February 2015.

   [I-D.johnston-rtcweb-media-privacy]
              Johnston, A. and P. Zimmermann, "RTCWEB Media Privacy",
              draft-johnston-rtcweb-media-privacy-00 (work in progress),
              May 2011.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [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, May 2010.

   [RFC6189]  Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
              Path Key Agreement for Unicast Secure RTP", RFC 6189,
              April 2011.

   [RFC6982]  Sheffer, Y. and A. Farrel, "Improving Awareness of Running
              Code: The Implementation Status Section", RFC 6982, July
              2013.

   [SJCL]     "Stanford JavaScript Crypto Library", SJCL
              https://bitwiseshiftleft.github.io/sjcl/, 2015,
              <https://bitwiseshiftleft.github.io/sjcl/>.

   [TADHACK]  "Telecom Application Developer Hackathon", TADHack 2015
              http://www.tadhack.com/2015/, 2015,
              <http://www.tadhack.com/2015/>.

   [TADHACK-ZRTP]
              "Telecom Application Developer Hackathon Remote Entry:
              WebRTC Security", TADHack 2015 Remote Entry
              https://www.youtube.com/watch?v=MOR2AYuRZ48, 2015,
              <https://www.youtube.com/watch?v=MOR2AYuRZ48>.








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   [WebRTC-API]
              Bergkvist, A., Burnett, D., Jennings, C., and A.
              Narayanan, "WebRTC 1.0: Real-time Communication Between
              Browsers", W3C Working Draft http://www.w3.org/TR/webrtc/,
              2013, <http://www.w3.org/TR/2012/WD-webrtc-20120821/>.

   [WebRTC-Book]
              Johnston, A. and D. Burnett, "WebRTC: APIs and RTCWEB
              Protocols of the HTML5 Real-Time Web", 3rd Edition,
              Digital Codex LLC, 2014, <http://webrtcbook.com>.

Authors' Addresses

   Alan Johnston
   Avaya
   St. Louis, MO
   USA

   Email: alan.b.johnston@gmail.com


   Phil Zimmermann
   Silent Circle
   Santa Cruz, CA
   USA

   Email: prz@mit.edu


   Jon Callas
   Silent Circle

   Email: jon@callas.org


   Travis Cross
   OfficeTone

   Email: tc@traviscross.com


   John Yoakum
   Avaya
   Cary, NC
   USA

   Email: yoakum@avaya.com




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