[asterisk-bugs] [JIRA] (ASTERISK-24646) PJSIP changeset 4899 breaks TLS

Stephan Eisvogel (JIRA) noreply at issues.asterisk.org
Sat Dec 27 15:42:34 CST 2014


Stephan Eisvogel created ASTERISK-24646:
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             Summary: PJSIP changeset 4899 breaks TLS
                 Key: ASTERISK-24646
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24646
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/Interoperability
    Affects Versions: 11.15.0
         Environment: Linux; hostile
            Reporter: Stephan Eisvogel


PJSIP as of changeset 4899 (https://trac.pjsip.org/repos/changeset/4899) has started verifying the Contact-header sent by the server to be of the SIPS scheme if transport is TLS. It will not check the Contact-header for ";transport=TLS" as sent by Asterisk.

As a result, registration by a client using this well-known stack will succeed, but any call attempt will terminate. A SIP trace will show the message "Warning: 381 localhost SIPS Required" going from the client to the server.

This was found using CSipSimple-trunk, other clients e.g. MicroSIP will likely follow, once this change has crept into their code bases.

The issue has previously been discussed last year here http://lists.digium.com/pipermail/asterisk-dev/2013-September/062567.html Asterisk developers were of the opinion that using SIPS in Contact-header will break proxying up a chain. PJSIP developers seem to be of the opinion they are following RFCs. And I am puzzled, looking for a resolution.

Workarounds/fixes I could identify:

1. Set disable_secure_dlg_check = PJ_TRUE on clients using PJSIP
2. Modify PJSIP's pjsip_inv_verify_request3 to check for ;transport=TLS not only in Record-Route-header but also in Contact-header.
3. Patch Asterisk to emit SIPS scheme when transport is TLS

I suggest identifying first, if this should be an Asterisk issue at all, or be brought up with PJSIP developers to change the default behaviour.



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