<div>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.</div><div><br></div><div>It will not work if you set it in the <global_settings> section.</div>
<div><br></div><div>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).</div>
<div><br></div><div>-Steve</div><div><br></div><div><br></div><div><br></div><div><div class="gmail_quote">On 6 May 2013 09:19, Farrukh Ali <span dir="ltr"><<a href="mailto:smfarrukh@live.com" target="_blank">smfarrukh@live.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr">I tried to configure rtp-timeout-sec in sofia.conf.xml as <br><pre><param name="rtp-timeout-sec" value="30"/></pre>but did not get any change, dead channels don't get hung up!<br>
Please reply<br><br>Regards,<br>Muhammad Farrukh<br><br><div><div></div><hr>Date: Fri, 3 May 2013 09:39:14 -0500<br>From: <a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a><div><div class="h5">
<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>Subject: Re: [Freeswitch-users] Dead channels in freeswitch<br><br>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. <br>
<div>On May 3, 2013 3:01 AM, "Steven Ayre" <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>> wrote:<br><blockquote style="border-left:1px #ccc solid;padding-left:1ex">
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.<div><br></div><div>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.</div>
<div><br></div><div>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).</div>
<div><div><br></div><div>To use session timers:</div><div><a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer" target="_blank">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer</a></div>
<div>
<a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout" target="_blank">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout</a></div><div><br></div><div>Or to use RTP timers:</div>
<div><a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec" target="_blank">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec</a></div>
<div><a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec" target="_blank">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec</a></div><div><br></div><div>-Steve</div>
<div><br></div>
<div><br></div><div><br><br><div>On 3 May 2013 07:12, Farrukh Ali <span dir="ltr"><<a href="mailto:smfarrukh@live.com" target="_blank">smfarrukh@live.com</a>></span> wrote:<br>
<blockquote style="border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr">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, <br>
And thanks for response<br><br>Regards<br>Muhammad Farrukh<br><br><div><div></div><hr>Date: Thu, 2 May 2013 08:51:42 -0400<br>From: <a href="mailto:bdfoster@davri.com" target="_blank">bdfoster@davri.com</a><div><div>
<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>Subject: Re: [Freeswitch-users] Dead channels in freeswitch<br><br><p dir="ltr">Just a note:</p>
<p dir="ltr">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. </p>
<p dir="ltr">-BDF</p>
<div>On May 2, 2013 7:09 AM, "Farrukh Ali" <<a href="mailto:smfarrukh@live.com" target="_blank">smfarrukh@live.com</a>> wrote:<br><blockquote style="border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr">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!<br>
and thanks to all again!<br><br>Regards,<br>Muhammad Farrukh<br><br><div><div></div><hr>From: <a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a><br>Date: Tue, 30 Apr 2013 17:20:36 +0100<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] Dead channels in freeswitch<br><br><div>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.</div>
<div><br></div><div>3 possibilities are:</div><div>1) Use session timers</div><div>2) There is a parameter to hang up a call is no RTP has been received for a certain period of time</div><div><span>3) Use SIP/TCP, keepalives might help (but the default is measured in hours)</span></div>
<div><span><br></span></div><div>Note that all of these are timeouts - you simply won't be able to know the call has ended immediately.</div><div><br>Steve</div><div><br></div><div><br></div><div><br>On 30 Apr 2013, at 08:33, Farrukh Ali <<a href="mailto:smfarrukh@live.com" target="_blank">smfarrukh@live.com</a>> wrote:<br>
<br></div><blockquote><div>
<div dir="ltr">Hi,<br><br>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<br>
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