Richard Shockey - NICC Standards

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Transcript Richard Shockey - NICC Standards

SHAKEN and STIRed: Thoughts on the Current State of:
Anti Spoofing / Caller Validation / Robocall Mitigation / Call Validation Display
Or “What are you insane Americans up to now?!”
NICC London 10 November 2016
Richard Shockey
Chairman SIP Forum
Shockey Consulting LLC
[email protected]
+1 703 593 2683
• Work in progress
• Disclaimer: The following opinions are those of a deranged, raving lunatic and do not necessarily reflect the opinions of the SIP
Forum or its member companies..
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Full Member Companies
(as of 11-05-2016)
How We Got Here
We wanted competitive voice markets. We got them, consequently no good deed goes
unpunished.
The Central issue now is restoring Trust in the Voice Networks.
Robocalls & Spoofing is the #1 complaint to the U.S. FCC and FTC.
https://consumercomplaints.fcc.gov/hc/en-us/articles/204009760-Consumer-Complaint-Charts-and-Data-Overview
U.S. Congress had held endless hearings. I was asked to testify.
https://energycommerce.house.gov/hearings-and-votes/hearings/modernizing-telephone-consumer-protection-act
Robocalls & Spoofing is the # 1 complaint to OFCOM and the UK ICO
https://ico.org.uk/action-weve-taken/nuisance-calls-and-messages/
I wrote one of the reports.
http://stakeholders.ofcom.org.uk/binaries/market-data-research/Ofcom_VoIP_RPKI_Report.pdf
Robocalls & Spoofing is the # 1 complaint to the CRTC in Ottawa
London Action Plan
http://londonactionplan.org/news/commitment-to-international-cooperation-london-action-plan-memberssign-mou/
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How We Got Here
Voice is still a $120 Billion dollar business across all the access platforms in the U.S. so there is
actually a business case to make this work.
Do the math. That means the UK Telecom Industry is trying to protect about 30 Billion Pounds of
annual revenue.
“A Billion Here, A Billion There.” It starts to add up to real money!
Shockey’s Law - “Money is the answer what is the question?”
The PSTN is undergoing a radical transition:
In the U.S. with VoLTE, SIP IMS IP-based voice will be 75% of the market in 3-4 years. Up from 35%
today. The U.S. is disconnecting the copper at a fast and furious pace.
Existing PSTN Class 5 TDM/SS7 equipment is at or near End of Life [EOL] and cannot be modified.
SIP is insecure and subject to all sorts of MiM attacks not to mention SBC’s.
TDM/SIP Gateways only complicate the security issue and are the biggest source of the problem.
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Robocalling/ Spoofing Timeline in the United States
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STIR & SHAKEN
The IETF STIR (Secure Telephone Identity Revisited) Working Group is developing a mechanism to allow phone
numbers to be “signed” at the origin, and “verified” at the termination. ATIS and the SIP Forum have proposed
enhancements to make this mechanism practical in the near-term by allowing service providers to perform the
“validation” and “verification” on the user’s behalf. Approval at the IETF is expected this year, and will set the stage
for the following additional steps:




SHAKEN: A United States service provider profile for STIR will provide implementation guidelines and
specify options within the protocol to ensure interoperability between all service providers. The
ATIS/SIP Forum IP-NNI Task Force will complete this profile framework (SHAKEN). Additional
enhancements to the profile will be developed in 2017.
Display Framework: A framework is required to allow for the display of validated Caller ID information
to end users in a consistent and secure format. The ATIS/SIP Forum IP-NNI Task Force is developing
this framework, with the initial deliverable expected by early 2017 .
Root Certificate Authority: the signing and verification of calling party information is based on
certificates issued by a recognized certificate authority. ATIS is working with the IP-NNI task force to
specify the technical requirements and operational procedures for the SHAKEN Root CA.
Testbed: ATIS is facilitating a testbed activity where they have developed a detailed test plan to
validate the SHAKEN framework. This will verify the protocol and ensure interoperability between
service providers. The STIR/SHAKEN Testbed is targeted to begin 4Q2016 and continue into 2017 and
beyond.
So What Happened? The US Strike Force on Robocalls
• https://www.fcc.gov/news-events/events/2016/10/second-meeting-industry-led-robocallstrike-force
• 30 Plus U.S. Carriers and the Supplier Community. 130 direct participants.
• Organized by Chairman Tom Wheeler of the FCC and Chaired by Randall Stephenson,
Chairman of the Board of AT&T
•
•
•
•
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Authentication Call Validation [ STIR / SHAKEN ]
Empowering Consumer Choice
Detection, Assessment, Traceback
Mitigation Regulatory Support/Root Cause Removal
What STIR – SHAKEN is Proposing
• We are going to cryptographically sign the SIP/IMS Call Signaling for every single call in the
U.S. network.
– Hopefully/Especially those coming from the International call gateways.
• STIR / SHAKEN use well-understood, well-deployed Public Key Infrastructure principals and
techniques. [PKI] X.509 Certificates & JWD Identity headers RFC 7519
– PKI is everywhere. Well-understood technology especially in Financial Services
• Private Cryptographic Credentials will be held by Originating Service Providers. Public
Cryptographic Keys will have to be distributed to Service Providers.
• Originating Service providers will make an attestation or “affirm” the information contained in
the SIP INVITE is true. That means the Caller ID among other data.
– If the Originating Service Provider cannot “affirm” the data in call then it MUST not sign the INVITE.
• The Terminating Service Provider will validate the claims in the INVITE and act accordingly.
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What will be Attested to..
• A. Full Attestation: The signing provider:



is responsible for the origination of the call onto the IP based service provider voice network
has a direct authenticated relationship with the customer and can identify the customer
has established a verified association with the telephone number used for the call.
Note: The legitimacy of the telephone number(s) the originator of the call can use is subject to
signer specific policy
• B. Partial Attestation: The signing provider:



is responsible for the origination of the call onto the telephone network
has a direct authenticated relationship with the customer and can identify the customer
has NOT established a verified association with the telephone number being used for the call
Note: Each customer will have a unique identifier, The unique identifier also provides a reliable
mechanism to identify the customer for forensic analysis or legal action where appropriate.
• C. Gateway Attestation: The signing provider:


is the entry point of the call onto the telephone network
has no relationship to the initiator of the call (e.g., international gateways).
Note: The signature will provide a unique identifier of the node. (The signer is not asserting
anything other than “this is the point where the call entered my network”.)
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Signaling Verification Now at 3GPP CT 1 & 3
Verstat Parameter
tel URI parameter in the P-Asserted-Identity
or FROM header field in a SIP requests
P-Asserted-Identity:
tel:+14085264000;verstat=TN-Validation-Passed
TN Validation Passes
TN Validation Failed
No TN Validation
Future: same values above for CNAM [Calling Name Delivery]
Security Considerations:
•
•
•
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The Verification Function must drop a verstat tel URI parameter received in an INVITE
If the terminating UE does not support the "verstat" parameter value, it must discard the
parameter
The terminating UE will act on the "verstat" parameter value, if the 200 (OK) response to
the UE REGISTER includes a Feature-Caps header field, as specified in RFC 6809°[190],
with a "+g.3gpp.verstat" header field parameter
STIR/SHAKEN ATIS/SIP Forum Call Flows for Call
Authentication / Verification
• It’s the last signaling hop we have
had concerns about. (5)
– After Call Validation has been
performed, what is the result
and then what does the
network or the consumer do?
• FCC has ruled we can block calls
with consumer consent.
• Can this be combined with
Enhanced CNAM?
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The Originating SIP Signaling Might Look Like This.
(This is what would go on the wire.)
• INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.36.78.177:60012;branch=z9hG4bK-524287-1---77ba17085d60f141;rport
Max-Forwards: 69
Contact: <sip:[email protected]:50207;rinstance=9da3088f36cc528e>
To: <sip:[email protected]>
From: "Test2"<sip:[email protected]>;tag=614bdb40
Call-ID: 79048YzkxNDA5NTI1MzA0OWFjOTFkMmFlODhiNTI2OWQ1ZTI
CSeq: 2 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE, OPTIONS
Content-Type: application/sdp
Date: Tue, 16 Aug 2016 19:23:38 GMT
Identity: lW84Z2BbPF8U4AWGg4eeKNlIYAq4j4KexICilTQJsfmEU23d2Nt7ih1valSKqwzXYctvJqsGzs5NuqAFgrLqg;info=<https://certauth.poc.sys.carrier.net/example.crt>;alg=ES256;canon=eyJ0eXAiOiJwYXNzcG9ydCIsImFsZyI
6IkVTMjU2IiwieDV1IjoiaHR0cHM6Ly9jZXJ0LWF1dGgucG9jLnN5cy5jb21jYXN0Lm5ldC9leGFtcGxlLmNl
cnQifQ.eyJkZXN0Ijp7InVyaSI6WyJzaXA6MTAwMEBzaXB0ZXN0LmNvbWNhc3QubmV0Il19LCJpYXQiOiIxND
cxMzc1NDE4Iiwib3JpZyI6eyJ1cmkiOiJzaXA6NTcxMjIyMzMzM0BzaXB0ZXN0LmNvbWNhc3QubmV0In19
Content-Length: 153
v=0
o=- 13103070023943130 1 IN IP4 10.36.78.177
c=IN IP4 10.36.78.177
t=0 0
m=audio 54242 RTP/AVP 0
a=sendrecv
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Call Analytics Call Flow at Termination
ANALYTICS
ENGINE
VERIFICATION
SERVICE
TERMINATING
CALL FLOW
TELEPHONY
APPLICATION
SERVER
CALL FLOW
USER
ENTITY
What Data Needs to be Carried in the INVITE to the User Agent / Entity
• The current consensus is carry the validation data from the last hop to the
Consumer/Enterprise User Entity via a set of new Call-INFO parameters. New IETF Drafts.
Note the author…
• https://datatracker.ietf.org/doc/draft-schulzrinne-dispatch-callinfo-spam/
• https://datatracker.ietf.org/doc/draft-schulzrinne-dispatch-status-unwanted/
• What data and how much data is required?
– Origin ? Confidence Level?
• Call-Info:
Termination_Report=<http://wwww.example.com/5974c8d942f120351143>
;purpose=info ;spam=85 ;type=fraud ;reason=“FCC_DNC list";text=”call
your mother”
•
• Data set must be useable by all clients.
• SHAKEN/STIR Highly Valid for National Security / Emergency Services Personnel
U.S. GETS etc. We have a very very good story to tell for them.
15 applications.
Enhanced Call Validation Display Options
[Good Call]
Existing User Display is limited to 15 Character ASCII
for CNAM [Calling Name Delivery] and the Calling
Party Number. This is what needs to be enhanced.
In mobile VoLTE, the handset is a SIP User Agent.
Now we can do anything!
Scenario 1
Calling party could display business name, address
and potentially a picture as well based on Enhanced
CNAM, but CNAM was not popular in the UK? Right?
Calling party can display alternative number to protect
Doctors privacy when responding to consumer
inquiries.
Protect Emergency Personnel from revealing their true
Calling Party Number.
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Enhanced Call Validation Display
[REALLY BAD Call]
Scenario 2
Network has no confidence in the signaling path
whatsoever; data analytics indicates possible
malicious call.
Signaling to consumer indicates very high level of
distrust in the call.
Network can alternatively block the call based on
clear consumer preference.
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Applicable to all SIP/IMS platforms
Cable today can optionally display Caller ID on
TV platforms. This could be added in.
A solution can work with any SIP-based
Enterprise PBX system either On Premise or
Hosted.
SIP Forum could take the lead there based on
our SIPconnect profiles.
Incorporation into 3GPP IMS.
We can’t fix POTS or TDM/SS7 nor do we
want to.
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STIR-SHAKEN / Strike Force / IETF /ATIS-SIP Forum / 3GPP
Multiple SDOs are still looking at various parts of this. It’s still a work in progress.
North American Carriers and providers using NANP [+1] numbers are considering
implementation strategies now.
• There is a strong desire/ demand /commitment to deploy.
• Suppliers are already asking serious questions on U.S. timelines and
roadmaps.
• ATIS and the SIP Forum will commit to Best Current Practices, Certificate
Management and Consumer Display Framework in 2017.
• Perhaps an Industry-Wide Agreement on visual Consumer Indicators.
IMHO the Call Validation process will not be effective unless consumers have some
sense of how the network judges the call session or the network can act in their
behalf.
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Other Issues to Be Resolved
This will take some time and the powers that be need to understand that.
• The solution needs to be tested.
– Carriers are looking into this. ATIS numbering testbed. SIP Forum SIPit.
– Effects on Post Dial Delay, etc.
• The issuance of PKI credentials for the Service Providers should
IMHO perfectly match the chain of authority for the Numbering
Plan itself. That begs the question of OTT providers.
• Security Reliability and Interoperability are obvious considerations.
• Consensus on Default PKI Encryption [EC256]?
• We need rings of defense. STIR/SHAKEN is not a Silver Bullet.
– Lots of other ideas out there. A lot of them really bad like blacklists.
– There is no flash cut here..
– Maybe direct support for enterprise PKI in the future.
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Issues for the U.S. Specifically to Resolve
How will the Certificate Trust Anchor be constituted?
By who? Under what governance and by what statutory authority?
How much will it cost?
There will have to be a Policy on who gets X 509 credentials and why.
The running theory in the U.S. is use of the NECA Operating Carrier Number [OCN] as well as direct
access to the NANP. Alternatively SIPD / Alt-SPID. It will be controlled.
Will all of this eventually be mandated by our regulator [FCC]?
I have heard the words “Notice of Inquiry [NOI] - Notice of Proposed Rule Making [NPRM]” –
Report and Order uttered in hushed tones.
Will the Carriers want cost recovery?
Implementing SHAKEN/STIR for Cable Operators is easy...not so easy for incumbents?
Are there privacy issues in inter-carrier data sharing and data analytics related?
Are there Legislative issues that need to be addressed?
Are there holes in “Authority to Act”?
The U.S. “Truth in Caller ID” Act is an oxymoron. Telephone Consumer Protection Act.
Proof of ‘intent’ to defraud is very difficult to prosecute.
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Interfaces.
We have 3 Classes of Membership: Full (Corporate) Members that
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Please see me about Full Membership!!
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First sign up as a Participant member of the SIP Forum and then join
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SIP Forum/ATIS NNI TF Landing page
http://www.sipforum.org/content/view/439/312/
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