[asterisk-bugs] [JIRA] (ASTERISK-21374) [patch] One-way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX

[David Brillert (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21374?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=206047#comment-206047 ] 

David Brillert commented on ASTERISK-21374:
-------------------------------------------

I held off upgrading to 11.4-rc1 due to the crash report.
So I never experienced the crash.
Hopefully jmls can test since he reported the issue.

I am glad to see you wrote a patch though since that means that soon I can finally upgrade to 11.4.0 or 11.5, or SVN (whenever this gets committed).  I have been holding off while waiting for this ticket to get closed.
                
> [patch] One-way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
> ------------------------------------------------------------------------------
>
>                 Key: ASTERISK-21374
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21374
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.3.0
>            Reporter: Michael L. Young
>            Assignee: Michael L. Young
>         Attachments: asterisk-21374-auto-nat-outgoing-fix.diff, asterisk-21374-fix-crash-and-rt-peers.diff, gdb.txt
>
>
> I found another case where the force_rport and comedia flags are not being set automatically when using the new auto_* settings.  This time it involves calls initiated by the PBX.
> When we reload asterisk the default flags turned on and off by auto_force_rport (force_rport) and auto_comedia (comedia) go back to the default setting of off.  These flags are turned on, as needed, when a peer re-registers or initiates a call.  This would apply to even just having the default global setting "nat=auto_force_rport".
> Everything is good except in the following scenario:
> We reload Asterisk and the peer's registration has not expired.  We load in the default settings for the peer which turns force_rport and comedia back to off.  Since the peer has not re-registered or placed a call yet, they remain off.  We then initiate a call to the peer from the PBX.  The force_rport and comedia flags stay off.  If NAT is involved, we end up with one-way audio since we never checked to see if the peer is behind NAT or not.

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