Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre@gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali <smfarrukh@live.com> wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ 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

Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster@davri.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" <smfarrukh@live.com> wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre@gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali <smfarrukh@live.com> wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ 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 _________________________________________________________________________ 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

If you're on VoIP and an endpoint completely disappears (eg wifi down) then the is absolutely no way for the other endpoint to detect that. That goes whatever software you're using. It's impossible for the down endpoint to tell the other that it's down, since it's down. Lack of messages also isn't enough to know it's down - it's normal during most of the SIP call for there to be no SIP packets, and RTP is designed to cope with packet loss. Absolutely the only way to detect it is to spot when you haven't heard from them for a while and timeout. Using SIP/UDP that's either in signalling with session timers (send a SIP packet, wait for a reply, timeout if no response within N seconds), or in media with RTP timers (hangup call if no RTP received for N seconds). To use session timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Or to use RTP timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec> -Steve On 3 May 2013 07:12, Farrukh Ali <smfarrukh@live.com> wrote:

Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response

Regards Muhammad Farrukh

------------------------------ Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster@davri.com

To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch

Just a note:

In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently.

-BDF On May 2, 2013 7:09 AM, "Farrukh Ali" <smfarrukh@live.com> wrote:

Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again!

Regards, Muhammad Farrukh

------------------------------ From: steveayre@gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch

SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP.

3 possibilities are: 1) Use session timers 2) There is a parameter to hang up a call is no RTP has been received for a certain period of time 3) Use SIP/TCP, keepalives might help (but the default is measured in hours)

Note that all of these are timeouts - you simply won't be able to know the call has ended immediately.

Steve

On 30 Apr 2013, at 08:33, Farrukh Ali <smfarrukh@live.com> wrote:

Hi,

I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply

_________________________________________________________________________ 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

_________________________________________________________________________ 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

_________________________________________________________________________ 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 tried to configure rtp-timeout-sec in sofia.conf.xml as <param name="rtp-timeout-sec" value="30"/>but did not get any change, dead channels don't get hung up! Please reply Regards, Muhammad Farrukh Date: Fri, 3 May 2013 09:39:14 -0500 From: anthony.minessale@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Asterisk uses the same 2 methods. The difference is probably that one or both are set by default or are implicit from some other setting. On May 3, 2013 3:01 AM, "Steven Ayre" <steveayre@gmail.com> wrote: If you're on VoIP and an endpoint completely disappears (eg wifi down) then the is absolutely no way for the other endpoint to detect that. That goes whatever software you're using. It's impossible for the down endpoint to tell the other that it's down, since it's down. Lack of messages also isn't enough to know it's down - it's normal during most of the SIP call for there to be no SIP packets, and RTP is designed to cope with packet loss. Absolutely the only way to detect it is to spot when you haven't heard from them for a while and timeout. Using SIP/UDP that's either in signalling with session timers (send a SIP packet, wait for a reply, timeout if no response within N seconds), or in media with RTP timers (hangup call if no RTP received for N seconds). To use session timers:http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Or to use RTP timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-s... -Steve On 3 May 2013 07:12, Farrukh Ali <smfarrukh@live.com> wrote: Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster@davri.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" <smfarrukh@live.com> wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre@gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali <smfarrukh@live.com> wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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

yes, same problem is with me, it looks like we are missing something,

Date: Mon, 6 May 2013 01:40:35 -0700 From: mehroz.ashraf85@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch

Stucked at the same place! See here, http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-td6...

if there is anything common in your setup! I have been asked to take to logs of that , thats what i coudnt manage to do at yet! Looking forward for possible solution !

-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________ 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, So, should I put it in profile tag? From: steveayre@gmail.com Date: Mon, 6 May 2013 11:08:17 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Did you ensure that the parameter was set on the sofia profile? By default they're not contained in the sofia.conf.xml file but in the files it pulls in via the X-PRE-PROCESS include. It will not work if you set it in the <global_settings> section. Also did you try restarting FS after setting the parameter (restarting the profile should also work, but restarting the process if you can is a simple way to be 100% sure it's reloaded the entire configuration). -Steve On 6 May 2013 09:19, Farrukh Ali <smfarrukh@live.com> wrote: I tried to configure rtp-timeout-sec in sofia.conf.xml as <param name="rtp-timeout-sec" value="30"/>but did not get any change, dead channels don't get hung up! Please reply Regards, Muhammad Farrukh Date: Fri, 3 May 2013 09:39:14 -0500 From: anthony.minessale@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Asterisk uses the same 2 methods. The difference is probably that one or both are set by default or are implicit from some other setting. On May 3, 2013 3:01 AM, "Steven Ayre" <steveayre@gmail.com> wrote: If you're on VoIP and an endpoint completely disappears (eg wifi down) then the is absolutely no way for the other endpoint to detect that. That goes whatever software you're using. It's impossible for the down endpoint to tell the other that it's down, since it's down. Lack of messages also isn't enough to know it's down - it's normal during most of the SIP call for there to be no SIP packets, and RTP is designed to cope with packet loss. Absolutely the only way to detect it is to spot when you haven't heard from them for a while and timeout. Using SIP/UDP that's either in signalling with session timers (send a SIP packet, wait for a reply, timeout if no response within N seconds), or in media with RTP timers (hangup call if no RTP received for N seconds). To use session timers:http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Or to use RTP timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-s... -Steve On 3 May 2013 07:12, Farrukh Ali <smfarrukh@live.com> wrote: Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster@davri.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" <smfarrukh@live.com> wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre@gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali <smfarrukh@live.com> wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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

Hey Steven, Thank you very much It works, Freeswitch stills shows an ongoing call, but stops its RTP traffic which was my concern, Thanks again And Mehroz you should try this as well! Regards Muhammad Farrukh From: steveayre@gmail.com Date: Mon, 6 May 2013 14:47:04 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch <profile> <settings> <param name="rtp-timeout-sec" value="30"/> </settings> </profile> Note the settings tag within profile. The same goes for the other parameters previously mentioned. -Steve On 6 May 2013 12:41, Farrukh Ali <smfarrukh@live.com> wrote: Hi Steve, So, should I put it in profile tag? From: steveayre@gmail.com Date: Mon, 6 May 2013 11:08:17 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Did you ensure that the parameter was set on the sofia profile? By default they're not contained in the sofia.conf.xml file but in the files it pulls in via the X-PRE-PROCESS include. It will not work if you set it in the <global_settings> section. Also did you try restarting FS after setting the parameter (restarting the profile should also work, but restarting the process if you can is a simple way to be 100% sure it's reloaded the entire configuration). -Steve On 6 May 2013 09:19, Farrukh Ali <smfarrukh@live.com> wrote: I tried to configure rtp-timeout-sec in sofia.conf.xml as <param name="rtp-timeout-sec" value="30"/>but did not get any change, dead channels don't get hung up! Please reply Regards, Muhammad Farrukh Date: Fri, 3 May 2013 09:39:14 -0500 From: anthony.minessale@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Asterisk uses the same 2 methods. The difference is probably that one or both are set by default or are implicit from some other setting. On May 3, 2013 3:01 AM, "Steven Ayre" <steveayre@gmail.com> wrote: If you're on VoIP and an endpoint completely disappears (eg wifi down) then the is absolutely no way for the other endpoint to detect that. That goes whatever software you're using. It's impossible for the down endpoint to tell the other that it's down, since it's down. Lack of messages also isn't enough to know it's down - it's normal during most of the SIP call for there to be no SIP packets, and RTP is designed to cope with packet loss. Absolutely the only way to detect it is to spot when you haven't heard from them for a while and timeout. Using SIP/UDP that's either in signalling with session timers (send a SIP packet, wait for a reply, timeout if no response within N seconds), or in media with RTP timers (hangup call if no RTP received for N seconds). To use session timers:http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Or to use RTP timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-s... -Steve On 3 May 2013 07:12, Farrukh Ali <smfarrukh@live.com> wrote: Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster@davri.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" <smfarrukh@live.com> wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre@gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali <smfarrukh@live.com> wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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 _________________________________________________________________________ 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

Hey Mehroz, The problem with this timer is it only STOPS the RTP traffic going from FS to SIPClient by detecting RTP is not received from the SIPClient for certain time(you set), It doesn't kill the session as you can check it by pressing F4, you need to hupall all the time, In Wireshark you could see that RTP stream is stopped after the timer expires! Regards Muhammad Farrukh

Date: Mon, 6 May 2013 23:15:18 -0700 From: mehroz.ashraf85@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch

I have already got that at the right place Farrukh!

<profile name="internal"> <settings>

</settings> </profile>

But unable to get the desired result! calls does not hangs up , which i monitored for 5 min and have to do "hupall" to get it fixed.

Is this parameter have any dependency over any other parameter , liker timers? as i have disabled session timer param name="enable-timer" value="false", but AFAIK session timer and rtp-timers are two different entities!

-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

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Is this master or stable, its not out of the question the param has a bug in master since there were some major changes and some media related params may need testing. The easy way would be lab it up and file a Jira so it can be quantified and resolved. On May 7, 2013 7:28 AM, "Steven Ayre" <steveayre@gmail.com> wrote:

No, it should actually hang up the call...

Steve

On 7 May 2013, at 10:12, Farrukh Ali <smfarrukh@live.com> wrote:

Hey Mehroz,

The problem with this timer is it only *STOPS *the RTP traffic going from FS to SIPClient by detecting RTP is not received from the SIPClient for certain time(you set), It doesn't kill the session as you can check it by pressing F4, you need to hupall all the time, In Wireshark you could see that RTP stream is stopped after the timer expires!

Regards Muhammad Farrukh

...

Date: Mon, 6 May 2013 23:15:18 -0700 From: mehroz.ashraf85@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch

I have already got that at the right place Farrukh!

<profile name="internal"> <settings>

</settings> </profile>

But unable to get the desired result! calls does not hangs up , which i monitored for 5 min and have to do "hupall" to get it fixed.

Is this parameter have any dependency over any other parameter , liker timers? as i have disabled session timer param name="enable-timer" value="false", but AFAIK session timer and rtp-timers are two different entities!

-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________ 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

_________________________________________________________________________ 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

Guessing this is related to the move of the media handling to crtp? -Steve On 7 May 2013 17:20, Anthony Minessale <anthony.minessale@gmail.com> wrote:

too slow,

I found the problem in master, if you don't want to update to get the fix, use the variable version of the param instead by setting rtp_timeout_sec in vars.xml

On Tue, May 7, 2013 at 8:48 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

...

Is this master or stable, its not out of the question the param has a bug in master since there were some major changes and some media related params may need testing. The easy way would be lab it up and file a Jira so it can be quantified and resolved.

On May 7, 2013 7:28 AM, "Steven Ayre" <steveayre@gmail.com> wrote:

...

No, it should actually hang up the call...

Steve

On 7 May 2013, at 10:12, Farrukh Ali <smfarrukh@live.com> wrote:

Hey Mehroz,

The problem with this timer is it only STOPS the RTP traffic going from FS to SIPClient by detecting RTP is not received from the SIPClient for certain time(you set), It doesn't kill the session as you can check it by pressing F4, you need to hupall all the time, In Wireshark you could see that RTP stream is stopped after the timer expires!

Regards Muhammad Farrukh

...

Date: Mon, 6 May 2013 23:15:18 -0700 From: mehroz.ashraf85@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch

I have already got that at the right place Farrukh!

<profile name="internal"> <settings>

</settings> </profile>

But unable to get the desired result! calls does not hangs up , which i monitored for 5 min and have to do "hupall" to get it fixed.

Is this parameter have any dependency over any other parameter , liker timers? as i have disabled session timer param name="enable-timer" value="false", but AFAIK session timer and rtp-timers are two different entities!

-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________ 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

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com

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commit ef5c1256f3a64502880bb232ea17a673e3e4eb10 Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Aug 8 14:46:38 2012 -0500 add rtp endpoint contributed by sangoma I'm guessing it's called crtp since there's a crtp_init function. -Steve On 7 May 2013 19:28, Michael Jerris <mike@jerris.com> wrote:

what is crtp?

On May 7, 2013, at 2:14 PM, Steven Ayre <steveayre@gmail.com> wrote:

...

Guessing this is related to the move of the media handling to crtp?

-Steve

On 7 May 2013 17:20, Anthony Minessale <anthony.minessale@gmail.com> wrote:

...

too slow,

I found the problem in master, if you don't want to update to get the fix, use the variable version of the param instead by setting rtp_timeout_sec in vars.xml

On Tue, May 7, 2013 at 8:48 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

...

Is this master or stable, its not out of the question the param has a bug in master since there were some major changes and some media related params may need testing. The easy way would be lab it up and file a Jira so it can be quantified and resolved.

On May 7, 2013 7:28 AM, "Steven Ayre" <steveayre@gmail.com> wrote:

...

No, it should actually hang up the call...

Steve

On 7 May 2013, at 10:12, Farrukh Ali <smfarrukh@live.com> wrote:

Hey Mehroz,

The problem with this timer is it only STOPS the RTP traffic going from FS to SIPClient by detecting RTP is not received from the SIPClient for certain time(you set), It doesn't kill the session as you can check it by pressing F4, you need to hupall all the time, In Wireshark you could see that RTP stream is stopped after the timer expires!

Regards Muhammad Farrukh

...

Date: Mon, 6 May 2013 23:15:18 -0700 From: mehroz.ashraf85@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch

I have already got that at the right place Farrukh!

<profile name="internal"> <settings>

</settings> </profile>

But unable to get the desired result! calls does not hangs up , which i monitored for 5 min and have to do "hupall" to get it fixed.

Is this parameter have any dependency over any other parameter , liker timers? as i have disabled session timer param name="enable-timer" value="false", but AFAIK session timer and rtp-timers are two different entities!

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Thanks Minessale , Setting up param name="rtp-timeout-sec" value="5" , in vars.xml , solved the issue. its working flawlessly! Thanks for all of yours contributions :) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.

Yes not in X-PRE-PROCESS, but with param name="rtp-timeout-sec" value="5" in vars.xml. I tested the scenario switching off the Wifi of both client.. it worked fine and call was terminated exactly after 5 sec. But today, i have seen another scenario, i.e, when one of the sip mobile app crashes down, it does not terminates that call. Either RTP is still in flow or something is misunderstood, However i can see the crashed app SIP device gone unregistered on freeswitch after ping option fails. !!!! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.

No, it'll be within the <settings> tag of the sofia <profile>. If you put it in vars.xml it'll be in completely the wrong place and mod_sofia won't see it... On 9 May 2013 08:29, mehroz <mehroz.ashraf85@gmail.com> wrote:

Yes not in X-PRE-PROCESS, but with param name="rtp-timeout-sec" value="5" in vars.xml.

I tested the scenario switching off the Wifi of both client.. it worked fine and call was terminated exactly after 5 sec.

But today, i have seen another scenario, i.e, when one of the sip mobile app crashes down, it does not terminates that call. Either RTP is still in flow or something is misunderstood, However i can see the crashed app SIP device gone unregistered on freeswitch after ping option fails. !!!!

-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.

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Ah ok :) hadn't tried it there Steve On 9 May 2013, at 19:14, Anthony Minessale <anthony.minessale@gmail.com> wrote:

rtp_timeout_sec in vars.xml will work but it will be global the profile param one will work but only if you update cos I found a bug in it.

On Thu, May 9, 2013 at 4:01 AM, Steven Ayre <steveayre@gmail.com> wrote:

...

No, it'll be within the <settings> tag of the sofia <profile>. If you put it in vars.xml it'll be in completely the wrong place and mod_sofia won't see it...

On 9 May 2013 08:29, mehroz <mehroz.ashraf85@gmail.com> wrote:

...

Yes not in X-PRE-PROCESS, but with param name="rtp-timeout-sec" value="5" in vars.xml.

I tested the scenario switching off the Wifi of both client.. it worked fine and call was terminated exactly after 5 sec.

But today, i have seen another scenario, i.e, when one of the sip mobile app crashes down, it does not terminates that call. Either RTP is still in flow or something is misunderstood, However i can see the crashed app SIP device gone unregistered on freeswitch after ping option fails. !!!!

-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.

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