sending BYE due to ACK timeout
Hi, My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio . The audio gets cut off after around 30 seconds, I think it detects RTP timeout. It's wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off. Is there any solution for this? Im using freeswitch 1.5 btw. Thanks, Jason
Its sending the bye because it is not seeing the ACK... This means the invite dialog has failed and it will end the call... You need to address the underlying NAT issue here On 1/19/14 10:27 AM, "Jason Sia" <jsia18@gmail.com> wrote:
Hi,
My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio . The audio gets cut off after around 30 seconds, I think it detects RTP timeout. It's wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off. Is there any solution for this? Im using freeswitch 1.5 btw.
Thanks, Jason
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH
Searching around the net you will find that usually the client does not handle Nat situation well - they put local IP in the sip contact. FS has a work around that using <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> Or maybe better in many cases <variable name="sip-force-contact" value="NDLB-connectile-dysfunction-2.0"/> that should work for you. On Mon, Jan 20, 2014 at 2:27 AM, Jason Sia <jsia18@gmail.com> wrote:
Hi,
My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio . The audio gets cut off after around 30 seconds, I think it detects RTP timeout. It's wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off. Is there any solution for this? Im using freeswitch 1.5 btw.
Thanks, Jason
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Steve Kieu
I will try to fix the NAT, will the auto nat feature fix this? On Mon, Jan 20, 2014 at 7:45 AM, Steve Kieu <msh.computing@gmail.com> wrote:
Searching around the net you will find that usually the client does not handle Nat situation well - they put local IP in the sip contact. FS has a work around that using
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
Or maybe better in many cases
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction-2.0"/>
that should work for you.
On Mon, Jan 20, 2014 at 2:27 AM, Jason Sia <jsia18@gmail.com> wrote:
Hi,
My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio . The audio gets cut off after around 30 seconds, I think it detects RTP timeout. It's wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off. Is there any solution for this? Im using freeswitch 1.5 btw.
Thanks, Jason
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Steve Kieu
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Hi Steve, I tried doing the setting in default.xml. I got the same experience. Thanks, Jason On Mon, Jan 20, 2014 at 8:47 AM, Jason Sia <jsia18@gmail.com> wrote:
I will try to fix the NAT, will the auto nat feature fix this?
On Mon, Jan 20, 2014 at 7:45 AM, Steve Kieu <msh.computing@gmail.com>wrote:
Searching around the net you will find that usually the client does not handle Nat situation well - they put local IP in the sip contact. FS has a work around that using
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
Or maybe better in many cases
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction-2.0"/>
that should work for you.
On Mon, Jan 20, 2014 at 2:27 AM, Jason Sia <jsia18@gmail.com> wrote:
Hi,
My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio . The audio gets cut off after around 30 seconds, I think it detects RTP timeout. It's wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off. Is there any solution for this? Im using freeswitch 1.5 btw.
Thanks, Jason
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Steve Kieu
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
I will try to fix the NAT, will the auto nat feature fix this?
auto-nat would automatically detect client behind nat and set the ext-rtp ip etc correctly. But it can not tell the client to set the sip contact header right, it is client bug (notably linphone on desktop linux did it as I saw) the best work around as I already posted is in FS (directory.xml for example add <variable name="sip-force-contact" value="NDLB-connectile-dysfunction-2.0"/>
On Mon, Jan 20, 2014 at 7:45 AM, Steve Kieu <msh.computing@gmail.com>wrote:
Searching around the net you will find that usually the client does not handle Nat situation well - they put local IP in the sip contact. FS has a work around that using
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
Or maybe better in many cases
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction-2.0"/>
that should work for you.
On Mon, Jan 20, 2014 at 2:27 AM, Jason Sia <jsia18@gmail.com> wrote:
Hi,
My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio . The audio gets cut off after around 30 seconds, I think it detects RTP timeout. It's wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off. Is there any solution for this? Im using freeswitch 1.5 btw.
Thanks, Jason
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Steve Kieu
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Steve Kieu
participants (3)
-
Jason Sia
-
Ken Rice
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Steve Kieu