<div dir="ltr"><span style="font-family:arial,sans-serif;font-size:13px">rtp_timeout_sec in vars.xml will work but it will be global</span><br><div><span style="font-family:arial,sans-serif;font-size:13px">the profile param one will work but only if you update cos I found a bug in it.</span></div>
<div><span style="font-family:arial,sans-serif;font-size:13px"><br></span></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, May 9, 2013 at 4:01 AM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">No, it'll be within the <settings> tag of the sofia <profile>. If you<br>
put it in vars.xml it'll be in completely the wrong place and<br>
mod_sofia won't see it...<br>
<div class="HOEnZb"><div class="h5"><br>
<br>
<br>
On 9 May 2013 08:29, mehroz <<a href="mailto:mehroz.ashraf85@gmail.com">mehroz.ashraf85@gmail.com</a>> wrote:<br>
> Yes not in X-PRE-PROCESS, but with<br>
> param name="rtp-timeout-sec" value="5" in vars.xml.<br>
><br>
> I tested the scenario switching off the Wifi of both client.. it worked fine<br>
> and call was terminated exactly after 5 sec.<br>
><br>
> But today, i have seen another scenario, i.e, when one of the sip mobile app<br>
> crashes down, it does not terminates that call. Either RTP is still in flow<br>
> or something is misunderstood, However i can see the crashed app SIP device<br>
> gone unregistered on freeswitch after ping option fails. !!!!<br>
><br>
><br>
><br>
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