[Freeswitch-users] RTCP Keep Alive issue - hangup after 60 seconds of silence
Christopher Rienzo
cmrienzo at gmail.com
Fri Jan 14 15:49:08 MSK 2011
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 at 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110114/96d90376/attachment.html
More information about the FreeSWITCH-users
mailing list