I don't know what RTCP keep alive is, but if they just mean to turn on RTCP, you can do it with the following params in your sofia configuration:
 <!-- enable rtcp on every channel also can be done per leg basis 

      with rtcp_audio_interval_msec variable set to passthru to pass 
      it across a call -->
 <param name="rtcp-audio-interval-msec" value="5000"/>
or, set the rtcp_audio_interval_msec channel variable.

See http://wiki.freeswitch.org/wiki/RTCP




On Fri, Jan 14, 2011 at 5:16 AM, Andy Ayers <andy@fabulous4.co.uk> wrote:

Hi,

 

Have encountered a problem recently where the company responsible for forwarding sip calls to my switch has started hanging up calls when there is more than 60 seconds of silence on one side of the call. As my service is primarily used for recording incoming messages, this means that any message being recorded longer than 60 seconds gets cut off. My provider says I need to configure freeswitch to send rtcp keep-alive packets to prevent them from hanging up the call.

 

Can anyone tell me how I can do this, I’ve checked the docs and can’t seem to find the right setting?

 

Many thanks

Andy


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