Re: draft-kuthan-sip-derive
Jiri Kuthan <[email protected]>
| Newsgroups | gmane.ietf.sip |
|---|---|
| Message-ID | <[email protected]> |
Hi Adam, thank you very much -- that's been most helpful indeed and I agree with pretty much all the points. I would highlight the scope and the actual verification mechanisms. I think that the scope, better-than-nothing, which just works with the same infrastructure, as for INVITE delivery, and makes on particular trust assumptions, begins to be agreeable. I agree too the actual verification mechanism, is difficult to agree upon. For now,I would resort to listing how such can differ from each other: - it it E2E (i.e., without any support in the network) -- that's obviously the objective - does it work with existing UA implementations, at least with some? That's desirable, not sure if really feasible. - if it leans on some existing SIP extensions, are these specified in a way that the scenario can unambiguously lean upon? (unfortunately not the case with dialog package) - does it work for requests that don't establish dialogs (preferably yes) - can it deal with forking? So far I have heard further suggestions for using some combination of OPTIONS, dialog authorization, or even a new method -- but we haven't really yet tested those hard. Also the applicability is a key aspect -- reverse routability agreeably being the most important one (apart from latency, security assumptions such as uncompromised DNS). I think the general approach is to explain that positive assertions are useful, negative don't give much, and document examples of when it fails, possibly accompanied by remedy suggestions: - unregistered UAC, UAC with call forwarding, anonymous UAC: don't do it :) (possibly offer to retry) - B2BUA: if you break it you fix it - tel URI .... perhaps resolving to IP if possible? - shared URI .... - UAC's proxy forwarding SUBSCRIBEs somewhere else... - 3pcc... -jiri Adam Roach wrote: > I've spent quite a bit of time over the past few days pondering the > approach laid out in the DERIVE draft. I think the approach of using > some form of "dial-back" to confirm identity has some merit (with a few > very strong caveats). I think the idea of leveraging already-defined > behavior is a good idea and somewhat clever; however, I suspect that the > exact mechanism defined in the current draft is not yet well enough > deployed for us to accept the limitations that it imposes. > > > The Need For User-Visible Caveats > ================================= > > As we've seen with the XMPP system, dial-back approaches can be used > with a fair amount of success to prevent certain types of identity > spoofing (see XEP-0220). Caveats associated with this approach in > general include reliance on security of the DNS and IP routing systems. > We have seen real-world attacks on bot h in the past [1][2], so we need > to make sure that user expectations for any dial-back system are set > appropriately. > > > SIP and Return Routability of AORs > ================================== > > However, when the flexibility of SIP routing is thrown into the mix, the > ability to reliably route back to the calling user's specific device > using that user's AOR becomes a much larger issue. > > For example, a caller at a call center my well advertise a calling > identity that reflects the contact point for that call center as their > "From" identity. However, calls made to that identity will either reach > an IVR or be dispatched to a random call center device that has no > knowledge of the calling party's dialog. Other normal subscribed > services, such as time-of-day routing, can have similar results. > Further, interaction with find-me-follow-me services -- or, really, any > service with serial forking -- can potentially result in dramatic > post-dial-delays that make execution of such a call-back service > infeasible. > > All of which is to say: any dial-back attempt using an AOR may > legitimately reach a device (or several devices) without reaching the > calling party's actual device. I would therefore assert that any > dial-back based strictly on AOR can at best confirm the veracity of the > "To" header field (with the DNS and routing caveats I mention above). It > cannot, under any circumstances, *refute* a claim. > > > Dialog-Package Re-Use > ===================== > > The actual use of the dialog event package in the DERIVE draft is > clever. The idea of being able to deploy a system based on what is > already in the field is very attractive, as it allows us to put the > solution into play immediately. However, based on results at SIPits, it > is supported by somewhere less than 50% of current UA implementations > (the estimate I got was "probably about one out of three"). > > That's actually not a bad percentage, especially considering that the > incidence of dialog package implementation should only go up -- it has > immense value in a number of useful services. However, the behavior that > the dialog event package needs to exhibit to support DERIVE is not > sufficiently well defined in RFC 4325. We've had a number of SIP experts > weigh in on the SIP mailing list with widely differing opinions about > how UAs *should* operate. And if *we* can't figure it out, I can > guarantee that implementors have done... let's use the work > "innovative"... things in this space. > > In other words, I fear that whatever gains we achieve by re-using the > dialog event package are largely negated by the fact that we're using > the very aspects of it that have been most poorly specified. > > > Potential Alternate Approaches > ============================== > > If we don't re-use the dialog event package as-is, then we need to > either find some other widely-deployed, well-defined UA behavior that we > can leverage, or we need to define new behavior on both the caller and > callee equipment. I can't immediately think of a solution that falls > into the former category -- perhaps someone else can come up with > something clever. That leaves defining some new mechanism to address a > call-back mechanism. Jiri has suggest defining a new method for > call-back validation, and I think this is a reasonable approach. > > Regardless of the approach taken, I would suggest that we should > leverage the use of public GRUUs to assist with routing to the proper > device. Devices that validate the identity of an incoming request can > render the public GRUU (with the ;gr parameter stripped) as the calling > party identity after they use it to reach the exact calling device. With > GRUUs thrown into the mix, we get around the issue of potentially being > unable to reach the calling party devices -- this allows us to return > actual negative hits in addition to positive hits. > > In fact, if the calling party is using a GRUU, we might be able to > achieve some interesting results by taking the GRUU out of the Contact > header-field of the INVITE and sending back an innocuous in-dialog > request of some kind (UPDATE? OPTIONS? Dare I say INFO?) *without* any > route headers to see whether we get a 481 response... > > > Conclusion > ========== > > In short, I support work towards dial-back "better than nothing" style > security. And, while I commend the authors of DERIVE for their > ingenuity, I fear that the dialog event package is a poor enough fit > that we should explore other alternatives before going too far down this > path. > > /a > > > [1] > http://arstechnica.com/news.ars/post/20080225-insecure-routing-redirects-youtube-to-pakistan.html > > [2] > http://arstechnica.com/news.ars/post/20071212-dns-poisoning-used-to-redirect-unwitting-surfers.html > > _______________________________________________ > Sip mailing list https://www.ietf.org/mailman/listinfo/sip > This list is for NEW development of the core SIP Protocol > Use [email protected] for questions on current sip > Use [email protected] for new developments on the application of sip > _______________________________________________ Sip mailing list https://www.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use [email protected] for questions on current sip Use [email protected] for new developments on the application of sip