[asterisk-bugs] [JIRA] (ASTERISK-24306) no direct media if srtp is used and thus higher cost for hosted PBX solution

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24306?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan closed ASTERISK-24306.
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    Resolution: Not A Bug

> no direct media if srtp is used and thus higher cost for hosted PBX solution
> ----------------------------------------------------------------------------
>
>                 Key: ASTERISK-24306
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24306
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: 11.8.1
>         Environment: any
>            Reporter: herman joossen
>
> I'm trying to use SRTP between 2 SNOM 710 phones, with FreePBX (2.11) in the middle.
> Without SRTP, both SNOM phones have direct media between each other after REINVITE was initiated by Asterisk. But this REINVITE is not initiated anymore as soon as SRTP is used so in this case Asterisk B2BUA stays in the middle and tha's not what I want.
> Both SNOM phones have identical codec settings, and in FreePBX their extensions are both programmed with canreinvite=yes, no recording options, no nat, no dial commands.
> Advanced Freepbx Device settings have SIP encryption = yes and canreinvite (directmedia) = yes.



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