[secdir] Secdir last call review of draft-ietf-rtcweb-security-11

Nancy Cam-Winget via Datatracker <noreply@ietf.org> Tue, 30 April 2019 18:05 UTC

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Subject: [secdir] Secdir last call review of draft-ietf-rtcweb-security-11
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Reviewer: Nancy Cam-Winget
Review result: Has Nits

SECDIR review of draft-ietf-rtcweb-security-11

Reviewer: Nancy Cam-Winget
Review result: Ready with Nits

I have reviewed this document as part of the security directorate'sÊ
ongoing effort to review all IETF documents being processed by theÊ
IESG.ÊÊThese comments were written primarily for the benefit of theÊ
security area directors.ÊÊDocument editors and WG chairs should treatÊ
these comments just like any other last call comments.

This document defines the threat model and security analysis of WebRTC.
There are a few normative clauses to suggest requirements, I think the document
could benefit of having more normative requirements to make it a stronger
standards track document. Otherwise, it reads more as an informational document.

The following are general comments and suggestions (and editorial nits at the
end):

Section 3. Should it also be noted that as it (browser) has no purview to the
actual application running, attacks from the application layer can still occur
but is not in scope for WebRTC?

Section 4.1
- The conceptual model is a bit confusing, as I think Entity can refer to both
the webRTC server as well as the receiving client application?  The notion is
that the User is trusting the webRTC (client) application (Entity A) to send
the media to another user, but really to the receiving application (Entity B). 
Perhaps model 1 has a typo where the Òtalk toÓ should be Entity B?

- The paragraph following the conceptual models is a bit awkward.  If I
understand correctly, the intent is to state that the user believes Entity A
when attempting to call Entity B, Entity A (the client app) could in fact also
send it to Entity C?  Which is valid, but the writing is awkward as the AÕs,
BÕs and CÕs are not explicitly stated as such or at least from the end userÕs
viewpoint.

- ÒIn either case, all the browser is able to do is verify and check
authorization for whoever is controlling where the media goesÓ Is ÒwhoeverÓ
meant to be the webRTC application controlling media paths?  That should be
clarified

- ÒÉconsent to access local devices is largely orthogonal to consent to
transmitÉÓ It should be both consent to local devices and the deviceÕs
resources, unless you mean Òlocal devicesÓ as in camera and microphone; but I
think processes and stored or cached files (e.g. other resources) need to be
considered too. § - Òconsent to send network traffic is about preventing the
        userÕs browser from being used to attack its local networkÓ
This is true, though, I think you are inferring that the network traffic is
enforced to be encrypted, which IÕm not sure is always the case; so I would
think it is up to the browser to ensure that this is the case unless the
browserÕs policy has made an exception

Section 4.1.2.1  ÒÉbug my computerÓ should be clarified.  I think there are at
least two dimensions to the ÒbugÓ, being it has ÒfreeÓ access to my resources
and it can actually potentially ÒlistenÓ to my calls; it could also potentially
override the use of my resources. The last sentence on the paragraph ÒNote that
question of consentÉ.Ó, the note makes sense, but I am not sure how the last
clause ÒÉ.the site is not listening inÓ, can you clarify this?

Section 4.1.2.2 ÒÉthe need for a second consentÓ, perhaps it is the need for a
ÒdistinctÓ consent as there may not be a previous consent?  By ÒdistinctÓ I
mean that it is a different type of consent than what may have been granted to
the car manuf (in your example) from getting my geolocationÉ.or perhaps I
missed the ÒfirstÓ consent type.

The last sentence of the paragraph is difficult to parse.  I think it is
asserting a requirement that the GUI used to launch the call must show the call
status (active, continuing, stopped)?

This section eludes to granting ÒcallÓ access only for the duration of the call
and the access should be limited (Òjust because I want some information on a
car doesnÕt meanÉ.Ó); the attack vector should be better highlighted.  As I
also believe that the browser can only grant the full client application
access, so for the duration of the call, it can very well be that the app can
get access to my resources beyond just the call (audio only vs. audio + videoÉ.)

Section 4.1.3
- The countermeasures can also be combined (e.g. consent is only given to a
given user for each call, as well as also having the appropriate keying
material). It is subtly eluded to in the 2nd to last paragraph but doesnÕt
consider that it can be done for all calls.

Section 4.1.4
- It should also be noted that weaknesses in the HTTPS stack can also be
exploited (weak authentication or key establishment use) by an attackerÉ.and
perhaps should be a MUST enforce strong mutual authentication and key
management.

Section 4.2.3
- As IÕm unfamiliar to ICE/STUN, IÕm not sure what checks are referred to in
ÒÉunsafe to completely remove the requirement for some checkÓ, this should be
clarified. Not sure (in the succeeding sentence) if there is a forward
reference to proposed checks or if they are listed elsewhere?

Section 4.3.1:  since the draft is listing requirements, ÒÉ.exchange mechanism
imperative forÉÓ
 The ÒimperativeÓ should be normative ÒMUSTÓ?

Section 4.3.2: who is the Òremote endpointÓ?

Editorial nits:
Section 1. It may be useful to describe why it is Òimmediately apparentÓ that
new security challenges resultÉ..suggest remove ÒimmediatelyÓ.

Section 4.1.1 (last sentence in the 2nd to last paragraph) is missing a toÕ:
Òsophisticated attack would be open up aÓ

Section 4.1.3 : ÒNow that we have seen another use caseÓ  Seems odd or
superfluous.  Suggest to just remove that clause or ÒWith the aforementioned
use cases, we can startÉ.Ó

Section 4.2.1: as it is a first reference, title should call out ÒInteractive
Connectivity Establishment (ICE)Ó, same for STUN (and STUNÕs reference, RFC
5389) should be called out.

Section 4.2.2 SRTP as a first reference should have its full reference to match
the acronym and TURN TCP should have reference too

Section 4.2.3 RTCP needs a reference?

Section 4.3: is it Òa problem from the SIP worldÓ or Òa problem familiar to the
SIP worldÓ? - Extra is Òcalling service is is non-maliciousÓ

Section 4.3.1: ÒinÓ should be removed ÒÉ.if end-to-end keying is in usedÓ

Section 4.3.2.1: ÒOne natural approach is to use É.Ó, natural approach to what?
 I think it is to mitigate the during-call attack only?