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OpenSIPit 2021 Notes


REMINDER: The Asterisk STIR/SHAKEN work was all done last year before things were finalized and there hasn't been anyone to test with until now so it's nor surprising that we have interoperability issues.

George's Notes:

  • We need to convert over to X509 certificates instead of EC certificates.
  • We need to send the "dest->tn" object in our Identity header.
  • We need to send a SIP "Date" header. Example from Liviu:
    • Wed, 14 Apr 2021 13:17:35 GMT
  • We're not filtering bad certificate URLs. The SipVicious guys were able to send a "file:///dev/random" URL which put Asterisk into a spin loop reading random data and writing it to /var/lib/asterisk/keys/stir_shaken/random.
  • We need to clarify and document the 3 types of certificate expiration...
    • We compare the current time to the timestamp sent in the Identity header and reject if there's more than signature_timeout seconds difference.
    • When we retrieve the certificate using the URL in the Identity header, we compare the HTTP "Cache-Control" "max-age" parameter or the Expires header to the time we cached the certificate the last time we retrieved it.
    • OpenSSL looks at the certificate valid date range.
  • Right now we fail if there's no "Cache-Control" or "Expires" header in the HTTP response we reject. That's not good.
  • We're using Base64 to encode and decode stuff and we should be using Base64URL instead.
  • We should be using randomly generated UUID for every request's orgid
  • We're saving certs in /var/lib/asterisk/keys/stir_shaken with the same name as in the URL. For instance: is saved as /var/lib/asterisk/keys/stir_shaken/cert.pem. If another request comes in like,( we overwrite Asterisk's cert.pem. We need to save the cert using the cert's serial number rather than its file name.
  • There are some error situations where the RFC says we MUST return a certain response code but we're leaving it to the dialplan author to check and reject and there's currently no way for the dialplan author to send specific SIP response codes. I suggest we do 2 things...
    • Create a pjsip dialplan function that can set a specific SIP response code for Hangup() to use.
    • Split the stir_shaken endpoint option to 2 options...
      • stir_shaken_attest = (yes | no) to control whether we send an Identity header or not
      • stir_shaken_verify = ( no | permissive | enforce ) where permissive would be the behavior we have today where we leave handling the verification result it up to the dialplan author and where enforcing makes res_pjsip_stir_shaken automatically send back the correct response code on failure.
  • We need enhance the STIR_SHAKEN dialplan function to add a verify_result_code code to get a numeric verification result code instead of the text result message.
  • We need to update the documentation for the STIR_SHAKEN function to include the possible result codes and messages.
  • When Asterisk is compiled with TEST_FRAMEWORK res_pjsip_stir_shaken doesn't do the check for the timestamp in the Identity header. It should.

Andreas Granig has a GitHub repo with a ton of Stir/Shaken sipp tests... It currently relies on a patched version of sipp (provided in the repo) but he's working to get the changes merged upstream. If anything, it's a good reference for what we should test ourselves.


George's Notes:

  • Eventually we (and pjproject) need to support the additional authentication digest algorithms defined in RFC8760 which include SHA-256 and SHA512-256.
  • We are not currently tolerant of RFC8760 compliant UASs that send us multiple WWW-Authenticate headers. If they send us one using SHA-256 and a second one using MD5, we fail when we see SHA-256 and don't continue to look for more WWW-Authenticate headers. I have a patch that at least makes us tolerant.