[Freeswitch-users] SRTP call between two extensions.

[Daniel Ivanov]

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 at 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
>>
>
>
> --
> Sent from mobile device
>
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