The rtptimeout feature has a few limitations:<br><br> . It is ineffective when the RTP is not terminated on Asterisk.<br><br>. It can cause false session hangups if the remote SIP UA does not support silence suppression <br>
<br>. The companion rtpholdtimeout can cause false hangups if you make incorrect judgment on how long a call hold can last.<br><br>. The rtptimeout period is not negotiated throughout the SIP signaling
path i.e. between the UAC, UAS, and intermediary proxies. So it does not help clear the session state throughout the network (when your BYE doesn't make it to all the entities in the SIP signaling path). <br><br>The SIP session-timers feature addresses all of the above limitations.
<br>
<br>--<br>Raj<br><br><br><br><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Jared,<br>I would think of using rtptimeout. There is a reason why you did not mention
<br>it and I am curious as to why.<br><div><div></div><div class="Wj3C7c"><br><br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">
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