which revision are you on?<br>The defaults on the latest code and examples should be configured to minimize delay.<br>Some of the older revisions built up some delay issues from udp buffering when timers were not synced.<br>
<br><br><div class="gmail_quote">On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <span dir="ltr"><<a href="mailto:pbd@suspiria.net">pbd@suspiria.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="purple" lang="DE">
<div>
<p><span lang="EN-US">When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay
?</span></p>
<p><span lang="EN-US"> </span></p>
<p><span lang="EN-US">I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.</span></p>
<p><span lang="EN-US"> </span></p>
<p><span lang="EN-US">Any advice?</span></p>
<p><span lang="EN-US"> </span></p>
<p><span lang="EN-US">Thanks</span></p>
</div>
</div>
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