You need to export rtp_secure_media=true
On Friday, January 10, 2014, Iskren Hadzhinedev wrote:
Hello, everyone!
I'm having trouble with SRTP calls between two internal extensions.
Whenever I configure the phones to require SRTP calls fail and the freeswitch console shows:
[INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
If I select "allow SRTP" so that encryption is not mandatory I get encryption on a single leg in a call.
Here's an example of what's happening:
Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success)
Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success)
Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail: INCOMPATIBLE_DESTINATION)
I've setup the sip_secure_media=true variable as instructed from
https://wiki.freeswitch.org/wiki/Secure_RTP
but it's still not working. Any help is greatly appreciated!
Kind regards,
--
Iskren Hadzhinedev