Archive for the 'SIP' Category

Hacking ZyXEL Gateways

Monday, March 24th, 2008 by

An interesting paper recently published by Adrian Pastor of ProCheckup discusses vulnerabilities and attacks against ZyXEL gateways, including (yikes) Remote wardriving/attacking internal networks over the Internet, among others:

  • Privilege escalation from “user‟ to “admin‟ account
  • SNMP read and SNMP write access enabled by default
  • Persistent XSS via SNMP
  • Poor session management allows hijacking of admin sessions
  • Authentication vulnerable to replay and password cracking attacks
  • Disclosure of credentials
  • Considering the code reuse among various products made by most vendors of these residential gateways, not to mention the widespread deployment by service providers, I think it would be quite interesting for VOIPSA folks to expand on Adrian Pastor’s work and pursue this type of testing on some of the VoIP gateway products that ZyXEL offers, specifically the Analog Telephone Adapter, Station Gateway and Integrated Access Device to start. Also, the web interface of embedded devices like these are especially problemmatic from a security perspective, and it’s well worth a look at another one of Adrian Pastor’s papers over at OWASP.

    “So what” you might say about the security of these types of devices? Well, SANS diary notes some strange things afoot at the Circle K with Dlink, and there is the recent BT Home Hub CVE-2008-1334 vulnerability. More routers and details at GNU Citizen’s router hacking challenge.

    Info on how to listen remotely to today’s RUCUS session at IETF

    Monday, March 10th, 2008 by

    ietflogo-1.jpgIf you are interested in listening in to today’s session here at IETF about “Reducing Unwanted Communications Using SIP” (RUCUS) which I’ve mentioned previously, I’ve posted information about how to participate in IETF remotely. The RUCUS session takes place from 1300-1500 US Eastern time today.

    Streaming audio should be available on ietf71-ch4.

    Jabber group chat should be available as well, but I don’t know yet in which chat room it will be. There isn’t yet a chat room on the IETF server for ‘rucus’. I’ll update this post once I know where the chat room is.

    UPDATE: A request is in to create the ‘rucus@jabber.ietf.org’ room. If that room isn’t created in time, we’ll use the SIPPING room at ‘sipping@jabber.ietf.org’. We’ll announce on the streaming audio which one we are using.

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    Web page for RUCUS BOF at IETF 71 now at new URL

    Monday, March 3rd, 2008 by

    ietflogo-1.jpgAs I mentioned previously (here and here), the “RUCUS” BOF about voice spam at IETF 71 in Philadelphia is one of great interest with its focus on voice spam, a.k.a. “SPam for Internet Telephony” or “SPIT”. Unfortunately BOF co-chair Hannes Tschofenig ran into a problem with his domain and had to move the page to a new URL: http://www.shingou.info/bof-rucus.html

    If you saved the URL or sent it on to someone, you’ll need to update to using the new URL. If you didn’t visit the RUCUS page before, please do check it out – and feel free to join the RUCUS mailing list. Of course, if you can, please do join us in person in Philadelphia!

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    Slides about Peer-to-peer SIP (P2PSIP) security now available

    Wednesday, February 20th, 2008 by

    ietflogo-1.jpgWant to learn more about the voip security aspects of peer-to-peer SIP? As I mentioned in the VOIPSEC mailing list last week, researchers from Huawei and the University of California recently released an Internet-Draft called “P2PSIP Security Analysis and Evaluation” which dives into an analysis of security issues in P2PSIP. It’s a good overview and one I’d strongly recommend to folks. (Note – you may want to read “P2PSIP Concepts” first to understand the language being used.)

    Beyond the Internet-Draft, though, the researchers announced yesterday that their slides are now available (PPT) that go into the issues. These are being prepared from presentation at the upcoming IETF 71 meeting March 10-14 in Philadelphia, so if you are attending the event you’ll be able to hear the presentation yourself.

    Peer-to-peer SIP is a fascinating area of current research and it’s good to see work like this being put into exploring the security aspects. Note – the researchers are looking for feedback so if you have comments on what you read, their contact information is in the Internet-Draft.

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    Join the new RUCUS mailing list if you want to look at ways to end SPIT!

    Friday, February 15th, 2008 by

    ietflogo.jpgAs mentioned previously, there is a new session planned for IETF 71 in March called “Reducing Unwanted Communications Using SIP“, a.k.a. “RUCUS”.

    The RUCUS mailing list is now open for subscriptions and we encourage anyone interested in looking at how we address the issue of voice spam, aka “Spam for Internet Telephony” aka “SPIT” to join into the conversation.

    We would ask you to please read the group description prior to joining so that you understand what we are trying to do. The primary goal of this session in March in Philadelphia is to look to understand the architecture necessary to address the issue and identify the pieces of that architecture that may already be there or may need to be put in place.

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    Cisco’s Slew of Vulnerabilities

    Thursday, February 14th, 2008 by

    Yesterday, Cisco released an advisory detailing a number of vulnerabilities which covered most recent versions of their IP phones, essentially the “Unified IP Phone” set of products, running both firmwares for SCCP and SIP. The covered vulnerabilities include a DNS Response Parsing Overflow, a Large ICMP Echo Request DoS, an HTTP Server DoS, an SSH Server DoS, a SIP MIME Boundary Overflow, a Telnet Server Overflow, and a SIP Proxy Response Overflow. Essentially, a wide range of vulnerabilities covering a number of the devices’ services and functionality.

    While it’s good that Cisco is actively taking steps to improve their products and are actually informing customers and the security community about the device’s security issues via security advisories, the scope and number of vulnerabilities involved in this one advisory seems to still be fairly indicative of the state of security for new VoIP products hitting the market, especially user agents and client devices. It would seem that as the rush continues for VoIP innovation and a quick to market product, much of these products’ security assessment due-diligence, not to mention many of their security features, are still being left in the dust…  Or at best, left for a firmware or software update post-launch.

    Blue Box Podcast Special Edition #23 – An Interview with Bob Bradley of Sonus Networks

    Tuesday, February 12th, 2008 by

    MD_bluebox157-2.jpgBlue Box Special Edition #23 is now available for download. In this podcast I sat down with Bob Bradley from Sonus Networks to talk about their products and solutions, how they secure customers networks and how they are different from other similar products in the market. I believe you’ll find it an interesting and useful introduction to the company.

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    End-to-end VoIP security using DTLS-SRTP? (A new proposal…)

    Monday, February 11th, 2008 by

    ietflogo.jpgAs we’ve discussed both here and on Blue Box, the issue of securing the keys for Secure RTP is one of the remaining challenges to have secure voice transmission in the open standards world of SIP. Out of the large number of proposals to secure the key exchange, “DTLS” emerged as the choice of the IETF… but it still had the issue that an endpoint needed to be sure of the authenticity of the other endpoint’s certificate. SIP Identity (RFC 4474) and a draft “Identity-Media” from Dan Wing addressed the authenticity issue but broke in some common network configurations. Now Kai Fisher has put out an Internet Draft called “End-to-End Security for DTLS-SRTP” that proposes a mechanism to address that. In the post to the SIP mailing list, Kai explains the motivation:

    I have submitted a draft proposing a solution to secure a DTLS-SRTP handshake and hence SRTP end-to-end (in terms of end-domain to end-domain). As discussed during the last IETF meetings and analyzed by Dan’s Identity-Media draft, current solutions like SIP Identity do not protect the authenticity of the fingerprint end-to-end in certain inter-domain scenarios. For example, a modification of SDP m-/c-lines or the From header field by intermediaries breaks the SIP-Identity or Identity-Media signature and causes a re-signing by a domain different to the originating one. The draft proposes a solution for such scenarios without the need to re-sign during domain traversal and which preserves the original identity information.

    The abstract to the draft provides more info:

    The end-to-end security properties of DTLS-SRTP depend on the authenticity of the certificate fingerprint exchanged in the signalling channel. In current approaches the authenticity is protected by SIP-Identity or SIP-Identity-Media. These types of signatures are broken if intermediaries like Session Border Controllers in other domains change specific information of the SIP header or the SIP body. The end-to-end security property between the originating and terminating domain is lost if these intermediaries re-sign the SIP message and create a new identity signature using their own domain credentials.

    This document defines a new signature type ‘Fingerprint-Identity’ which is exchanged in the signalling channel. Fingerprint-Identity covers only those elements of a SIP message necessary to authenticate the certificate fingerprint and to secure media end-to-end. It is independent from SIP-Identity and SIP-Identity-Media and can be applied in parallel to them.

    More details can, of course, be found in the draft. As noted in the post to the SIP mailing list, Kai is looking for feedback. This is an important issue to get done – and to get done correctly – so we strongly urge people to take a look at the document and provide feedback if you see ways the proposal can be improved.

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    Raising a RUCUS about SPIT at IETF 71!

    Monday, February 4th, 2008 by

    UPDATE: The RUCUS mailing list is now open for subscription.

    ietflogo.jpgWant to get together with others and discuss in further detail what we can do about Spam for Internet Telephony (SPIT)? A new session has been approved for the IETF 71 meeting coming up in Philadelphia in March called “Reducing Unwanted Communications using SIP” a.k.a. “RUCUS” (Hey, it’s not a real IETF group until it has a cute acronym!) Hannes Tschofenig, who submitted the proposal, has created a RUCUS web page and is looking for feedback. The page says in part:

    The topic of dealing with unwanted traffic in SIP has surfaced several times in the IETF in the context of preventing Spam for Internet telephony. Previous attempts to have a structured discussion about this topic have (among other reasons) failed due to the strong focus on selected solution approaches.

    Prior work in SIP on identity management has an important role in this activity since a strong identity mechanism in SIP has been seen as a prerequisity for establishing authorization policies. Hence, the “Discussion and Analysis of SIP Identity” (DASI) BoF is relevant for this event. Even though there is no direct dependency between the two activities the number of interested participants will quite likely overlap.

    This BoF focuses on the discussion of architectural aspects. The underlying theme is that the work on building blocks is more fruitful once the larger framework is understood. A number of solutions components have been submitted to the IETF, have been published in the academic literature and found their way into other standardization bodies. Reduce unwanted communication requires authorization decisions to be made. These decisions can be made based on individual sessions but also on the interaction at a higher granularity (e.g., the interaction with a specific VoIP provider network). Examples of questions with relevance for an architecture might be:

    - Where does information for decision making come from?

    - What are useful information items for decision making?

    - Where are policy decision points located? What about the placement of
    policy enforcement points?

    - Are privacy aspects to consider with the exchange of information?

    - How does the underlying trust model look like?

    - What assumptions are certain mechanisms based on?

    - Can individual proposals be combined in a reasonable way?
    etc.

    It is not the aim of the BoF to discuss specific solution approaches since it is likely that multiple techniques have to be used in concert.

    If you are attending IETF 71 in Philadelphia in March, do plan on joining in the RUCUS! (I’ll be there.)

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    VoiceCon Orlando to offer “SIP Security” talk

    Sunday, January 27th, 2008 by

    Over on his new No Jitter blog, Eric Krapf notes in his SIP Security post that at VoiceCon Orlando they will be running a SIP security talk again:

    “As SIP continues to seep into the mainstream, more attention is being paid to security issues, especially in public IP networks/the Internet. At VoiceCon Orlando in March, we’re bringing back Cullen Jennings and Eric Rescorla to once again give their ‘SIP Security’ tutorial, which offers enterprises a jump on many of the key issues.”

    Long-time readers will remember that I wrote about Cullen & Eric’s appearance at VoiceCon San Francisco back in August and I am glad to see they’ll be back again in Orlando. Since I’ll be down there at VoiceCon Orlando, I’ll look forward to seeing them both again (and yes, I’ll probably sit in their presentation again :-) .

    Eric also reviews a couple of the ETSI security presentations I recently mentioned, giving a better glimpse than I did here! :-)

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