[asterisk-bugs] [JIRA] (ASTERISK-26457) [patch] force_rport, auto_comedia: No NAT detection triggered.

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26457?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232635#comment-232635 ] 

Asterisk Team commented on ASTERISK-26457:
------------------------------------------

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].

> [patch] force_rport,auto_comedia: No NAT detection triggered.
> -------------------------------------------------------------
>
>                 Key: ASTERISK-26457
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26457
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.23.1, 13.11.2, 14.0.2
>            Reporter: Alexander Traud
>
> *Steps to Reproduce*
> # Asterisk with {{nat=force_rport,auto_comedia}}
> # first VoIP/SIP client (caller) uses public IP addresses in its SDP
>   for example not within a NAT, like IPv6
> # second VoIP/SIP client (callee) uses private IP addresses in its SDP
>   for example within a NAT, for example IPv4 without STUN
> # call is established = signaling via SIP is OK
> *Expected Results*
> Media (RTP) should flow, because comedia is enabled as Asterisk detected a NAT towards the callee. Asterisk is sending RTP to public IP addresses.
> *Actual Results*
> Media is one way (from callee to caller). Asterisk sends the media of the caller to the address mentioned in the SDP message of the callee. That was a private IP address. Therefore media does not reach the callee. Therefore one-way media.
> *Workaround*
> {{nat=auto_force_rport,auto_comedia}} fixed the issue for me, because the related code tests for a NAT in that case.
> Asterisk should test for NAT, whether {{auto_force_rport}} or {{auto_comedia}} is set. This is done in other calling scenarios within Asterisk already. The attached patch does this for this scenario here as well.



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