<br>being unreachable does not cause delay, sending the audio too fast does.<br>the timer in the rtp stack will slow down the audio stream to match the clocked rate and if there is a difference<br>it can translate to delay over time.<br>
<br>This is been especially prevalent in cisco phones but other things have caused a similar problem.<br><br>you can also try disabling rtp timer by setting rtp_timer_name=none before you answer the call.<br><br>the keys are controlled by the mappings in the conference profile <br>
<br><br>it's a little unnerving talking to someone named public dump, do you have an identity?<br><br><br><br><div class="gmail_quote">On Wed, Aug 26, 2009 at 4:36 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 style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">I am quite sure that it’s not a problem with the network connection, since the delay builds up gradually over time, while the host is reachable
without any problems.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">I have added the suggested config lines and will check the results.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">By the way, can I remap the hash key, so that is does not end the conference ? I’ve tried remapping it, but freeswitch seems to ignore the setting
(the console shows: hangup mapped to ‘*’ … but # still ends the call).</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">It is irritating for users, since most users try to enter the conference pin the #-key at the end.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<div style="border-style: none none none solid; border-color: -moz-use-text-color -moz-use-text-color -moz-use-text-color blue; border-width: medium medium medium 1.5pt; padding: 0cm 0cm 0cm 4pt;">
<div>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0cm 0cm;">
<p><b><span style="font-size: 10pt;">Von:</span></b><span style="font-size: 10pt;"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>]
<b>Im Auftrag von </b>Anthony Minessale<br>
<b>Gesendet:</b> Mittwoch, 26. August 2009 22:45<div><div></div><div class="h5"><br>
<b>An:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b>Betreff:</b> Re: [Freeswitch-users] delay buildup in conference</div></div></span></p>
</div>
</div><div><div></div><div class="h5">
<p> </p>
<p style="margin-bottom: 12pt;">There is no bug, it's all dependent on your network conditions.<br>
I spend literally 8-12 hours a day on a conference and there is no delay.<br>
<br>
The important param is <br>
<param name="rtp-autoflush-during-bridge" value="true"/><br>
<br>
in the sofia profile in question.<br>
<br>
if you have delay with that in place, then it's probably not FS<br>
<br>
</p>
<div>
<p>On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <<a href="mailto:pbd@suspiria.net" target="_blank">pbd@suspiria.net</a>> wrote:</p>
<div>
<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Hi,</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">I am on a quite recent version (i assume):</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">FreeSWITCH Version 1.0.trunk (14461)</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality).</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Thanks</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<div style="border-style: none none none solid; border-color: -moz-use-text-color -moz-use-text-color -moz-use-text-color blue; border-width: medium medium medium 1.5pt; padding: 0cm 0cm 0cm 4pt;">
<div>
<div style="border-style: solid none none; border-color: -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0cm 0cm;">
<p><b><span style="font-size: 10pt;">Von:</span></b><span style="font-size: 10pt;">
<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>]
<b>Im Auftrag von </b>Anthony Minessale<br>
<b>Gesendet:</b> Mittwoch, 26. August 2009 20:57<br>
<b>An:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">
freeswitch-users@lists.freeswitch.org</a><br>
<b>Betreff:</b> Re: [Freeswitch-users] delay buildup in conference</span></p>
</div>
</div>
<div>
<div>
<p> </p>
<p style="margin-bottom: 12pt;">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.</p>
<div>
<p>On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <<a href="mailto:pbd@suspiria.net" target="_blank">pbd@suspiria.net</a>> wrote:</p>
<div>
<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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