[asterisk-bugs] [JIRA] (ASTERISK-25242) PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT

[Mark Michelson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25242?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226846#comment-226846 ] 

Mark Michelson commented on ASTERISK-25242:
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The proposed solution here for PJSIP is to implement the same thing that chan_sip has for its "rtpkeepalive" option.

> PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-25242
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25242
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>            Reporter: Mark Michelson
>
> When Asterisk is behind a NAT and the two parties involved in the call are in front of the NAT, it is likely that there will be no audio on the call. The reason is that the NAT will not allow the inbound media to Asterisk through since Asterisk has not punch a hole through the NAT from the inside.
> There are workarounds that can be performed in the dialplan, such as playing some silence to one of the parties once the call is answered. This, however, is not very elegant, and it may not seem obvious to an Asterisk deployer if encountered.



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