Internet DRAFT - draft-johnston-rtcweb-zrtp
draft-johnston-rtcweb-zrtp
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
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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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