[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:comment-tabpanel&focusedCommentId=222573#comment-222573 ] 

Matt Jordan commented on ASTERISK-24306:
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Features requests are no longer submitted to or accepted through the issue tracker. Features requests are openly discussed on the mailing lists [1] and Asterisk IRC channels and made note of by Bug Marshals.

[1] http://www.asterisk.org/support/mailing-lists

For direct media to work with SRTP, something would have to exchange the keys between the phones. Before direct media is initiated, the entire call must first be set up. Given the following:
{noformat}
Phone A <=> Asterisk <=> Phone B
{noformat}
Asterisk would have to not only re-INVITE the media session between Phone A and Phone B, it would have to inform both phones that the SRTP session that was established between Asterisk and each phone was no longer valid. It would then have to forward the key information from both sides to the other phones, and hope that the phones support re-establishing the SRTP session in such a fashion.



> 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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