[Freeswitch-users] RTCP Keep Alive issue - hangup after 60 seconds of silence

Anthony Minessale anthony.minessale at gmail.com
Tue Jan 18 00:11:59 MSK 2011


if you want the easy way out

set

record_waste_resources=true

before you run the record app.  Then you will send rtp the whole time.


On Mon, Jan 17, 2011 at 1:52 PM, Andy Ayers <andy at fabulous4.co.uk> wrote:
> Many thanks for all your help. Should I be able to see the RTCP messages
> going out in the log file?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: 14 January 2011 15:00
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60
> seconds of silence
>
>
>
> While this will turn on RTCP your provider needs to be beaten for requiring
> such a resource wasting process.
>
>
>
> /b
>
>
>
> On Jan 14, 2011, at 6:49 AM, Christopher Rienzo wrote:
>
> 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
>
>
>
> _______________________________________________
> 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
>
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900



More information about the FreeSWITCH-users mailing list