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

[Mark Michelson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24646?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224359#comment-224359 ] 

Mark Michelson commented on ASTERISK-24646:
-------------------------------------------

So before I take any further action, I want to make sure I understand exactly what the problem is you're seeing. Here's how I'm interpreting the problem:

* CSipSimple registers to Asterisk.
* CSipSimple places a call over TLS to Asterisk. The Request URI of the incoming call to Asterisk has a sips: URI scheme.
* Asterisk sends a response to the INVITE with a Contact header that does not have a sips: URI scheme. Instead, the Contact header that Asterisk sends has a sip: URI scheme with ;transport=tls appended.
* CSipSimple, which uses PJSIP under the hood, checks the 200 OK and complains that the Contact header in the 200 OK from Asterisk is not a sips: URI, so after ACKing the 200 OK, CSipSimple sends a BYE to end the dialog.

Do I have the scenario correct, or does it go differently from that?

> 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
>            Assignee: Mark Michelson
>
> 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.



--
This message was sent by Atlassian JIRA
(v6.2#6252)