[asterisk-bugs] [JIRA] (ASTERISK-27001) PJSIP TLS connection not stable

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27001?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236998#comment-236998 ] 

Asterisk Team commented on ASTERISK-27001:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> PJSIP TLS connection not stable
> -------------------------------
>
>                 Key: ASTERISK-27001
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27001
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 13.15.0
>         Environment: centos 6.8(64-bit)
>            Reporter: Ian Gilmour
>
> Hi,
> I have a development Asterisk 13.15.0 test setup (uses the bundled pjsip-2.6).
> On startup Asterisk registers 1 Asterisk users with a remote OpenSIPS server, over TLS, using the PJSIP stack. As part of the test this Asterisk PJSIP user is reregistered with OpenSIPS Server every couple of mins.
> All outgoing/incoming pjsip call media is encrypted using SRTP and via an external RTPPROXY running alongside the external OpenSIPS Server.
> Asterisk is additionally configured to use PJSIP on 127.0.0.1:5060 to allow calls from a locally run SIPp process. All SIPp calls are TCP+RTP.
> I use SIPp to run multiple concurrent loopback calls (calls vary in
> duration) through Asterisk to the OpenSIPS server and back to an echo() service running on the same Asterisk).
> i.e.
> {noformat}
>   SIPp <-TCP/RTP-> Asterisk <-TLS/SRTP->
>       OpenSIPS server (+ rtpproxy) <-TLS/SRTP-> Asterisk (echo service).
> {noformat}
> With no calls running the PJSIP TLS connection stays up and I see it reregistering the user every ~2mins.
> When I start to run the SIPp test I start seeing the PJSIP stack having TLS issues - closing the current TCP port as a result, in this state outgoing SIPp calls obviously start failing. 
> A few seconds later Asterisk (PJSIP) opens a new port, reregistering with the OpenSIPS server, and the calls continue.
> If I switch Asterisk to use the chan_sip stack rather than the PJSIP stack for the TLS connection to the OpenSIPS server the connection stays up with no call failures.
> I patched a couple of PJSIP files to help me see what's going on and I have played with the PJSIP TLS code. I can improve the reliability of the connection by ignoring a specific error condition (see the code within #if EXPERIMENTAL...#endif in the attached patch).



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