[asterisk-bugs] [JIRA] (ASTERISK-25242) PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'?

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25242?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-25242:
------------------------------------

    Affects Version/s: SVN
                       11.18.0
                       13.4.0

> PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'?
> --------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-25242
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25242
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: SVN, 13.4.0
>            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 punched 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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