<div dir="ltr">I guess you are talking about latency issues caused by the jitter buffer.<div>Does the problem go away when you put this </div><div><br></div><div><action application="set" data="sip_jitter_buffer_during_bridge=false"/><br>
</div><div><br></div><div>in the dialplan?</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Nov 19, 2013 at 5:32 PM, I put the Who? in Mishehu <span dir="ltr"><<a href="mailto:mishehu@freeswitch.org" target="_blank">mishehu@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>I do not have this problem, and I use
FS for years. So this is not normal or expected behavior. But
that being said, you provided no information about your set up -
such as distro, FS version, what type of system you were running
on (virtual or bare metal), and what kind of network you were
running this over.<br>
<br>
I can only guess that you're running on a virtual server and are
experiencing some sort of clock skew, but that's just a guess.<br>
<br>
<pre cols="72">--
Yossi Neiman
On 11/19/2013 02:28 PM, Ashwin Jain wrote:
</pre>
</div>
<blockquote type="cite"><div><div class="h5">
<div dir="ltr">
<div>Hi,</div>
<div><br>
</div>
We have setup a single freeswitch server which receives a call
from a DID provider (ex: voipms) and do a outbound call another
provider (ex: lesnet) (we also record call).
<div><br>
</div>
<div>For short duration call, everything is working fine, but
for calls with long duration (like more than 10 minutes),
audio is getting out of sync.<br clear="all">
<div><br>
</div>
<div>Has anyone had this issue? Where could be the problem? </div>
<div><br>
</div>
-- <br>
<div dir="ltr">Thanks and Regards,<br>
Ashwin Jain<br>
</div>
</div>
</div>
<br>
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