<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><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 style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.296875); -webkit-composition-fill-color: rgba(175, 192, 227, 0.230469); -webkit-composition-frame-color: rgba(77, 128, 180, 0.230469); ">3) Use SIP/TCP, keepalives might help (but the default is measured in hours)</span></div><div><span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.296875); -webkit-composition-fill-color: rgba(175, 192, 227, 0.230469); -webkit-composition-frame-color: rgba(77, 128, 180, 0.230469); "><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">smfarrukh@live.com</a>> wrote:<br><br></div><blockquote type="cite"><div>
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<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> </div>
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