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

[herman joossen (JIRA)]

herman joossen created ASTERISK-24306:
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             Summary: 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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