Yeah, try kristian's suggestion first. At some point that variable changed. This happened to me last week and turned out my colleagues checked out the 1.5 branch instead of 1.2stable. 


On Fri, Jan 10, 2014 at 7:14 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
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



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