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,

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Iskren Hadzhinedev



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