Subject: [Sip] Removal of RTP-to-SRTP-upgrade requirement in draft-ietf-sip-media-security-requirements

From: "Schneider, Peter (NSN - DE/Muenich)" <[email-masked]>
Date: Mon, 3 Mar 2008 10:04:31 +0100

Former versions of the draft included the requirement

   R15:  The media security key management protocol SHOULD allow
         endpoints to start with RTP and then upgrade to SRTP.

The current version -03 doesn't contain the requirement.

As quoted from RFC3264 in the draft, after the INVITE with the SDP
offer, the offerer must be prepared to receive media, e.g. for
announcements. I feel it makes sense to support a use case where such
announcements can be sent unencrypted, even if the caller wants to set
up an encrypted call. Obviously, this allows an attack (e.g. sending of
forged announcements), but on the other hand, it may be useful for the
caller to hear the announcements. It could be a local policy of the
calling user agent to drop such media (in order to avoid attacks) or to
play it out. The user agent may also be configured to indicate somehow
whether "the call is secured" or not.

So I suggest to revive R15, maybe in a wording that makes more clear
what use case(s) should be supported.

Peter
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---

From: "Dan Wing" <[email-masked]>
Date: Tue, 11 Mar 2008 13:49:30 -0400

Thanks for your comment on the draft.

Brian Stucker's now-expired draft, "Coping with Early Media in the Session
Initiation Protocol",
http://tools.ietf.org/html/draft-stucker-sipping-early-media-coping-03,
describes several types of early media:

  Pre-routing early media  
  Pre-presentation early media 
  Post-presentation early media
  Non-SDP early media 

I'm not sure which of those we would want to use to fulfill requirement R15 --
they have quite different security properties from an attack that concerns me.
My concern is that R15 would put an SRTP caller at risk of receiving an
attacker's RTP.  Once the called party's SRTP key is known, I expect the
caller could discard the incoming RTP in favor of the SRTP.  But for a time
period (between the INVITE being sent and the called party's SRTP key being
known), the calling party is vulnerable to receiving (and playing out)
arbitrary RTP media from any arbitrary host on the network.

-d

> -----Original Message-----
> From: [email-masked] [mailto:[email-masked]] On 
> Behalf Of Schneider, Peter (NSN - DE/Muenich)
> Sent: Monday, March 03, 2008 4:05 AM
> To: [email-masked]
> Subject: [Sip] Removal of RTP-to-SRTP-upgrade requirement 
> indraft-ietf-sip-media-security-requirements
>