RTCP Keep Alive issue - hangup after 60 seconds of silence
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
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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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.
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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
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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
FYI, This variable: http://wiki.freeswitch.org/wiki/Variable_record_waste_resources Wasn't listed on the main chan vars page. I just rectified that by adding it to the call recording related variables section. -MC On Mon, Jan 17, 2011 at 1:11 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
All sorted, Many thanks! From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 January 2011 00:03 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60 seconds of silence FYI, This variable: http://wiki.freeswitch.org/wiki/Variable_record_waste_resources Wasn't listed on the main chan vars page. I just rectified that by adding it to the call recording related variables section. -MC On Mon, Jan 17, 2011 at 1:11 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: 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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (5)
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Andy Ayers
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Anthony Minessale
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Brian West
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Christopher Rienzo
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Michael Collins