If you're sure it's not the end points, than is traffic is passing thru a stateful firewall (either direction/either side) where the SFW's UDP state timers may be too fast. 

/S

—————
iThing:  Big thumbs & little keys. Please excuse typo, spelling and grammar errors • Thought of the Day – "With all this manure, there must be a pony in here somewhere.”

On 2013-05-03, at 2:17, 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
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_________________________________________________________________________
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
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http://wiki.freeswitch.org
http://www.cluecon.com

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_________________________________________________________________________ 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
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