1.6.3 should be rolling out today hopefully. This will be roughly what is in master now.<span></span><br><br>On Friday, November 20, 2015, Dragos Oancea <<a href="mailto:doancea@orange-vallee.net">doancea@orange-vallee.net</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>
<div text="#000000" bgcolor="#ffffff">
Hi<br>
<br>
Josh, FS master is not 1.6.2 , it's the latest master branch . Get it
from git and compile it.<br>
1.6.2 does not have the new JB.<br>
<br>
I assume you don't transcode at all and you use G.711 along the path.<br>
Try using ILBC between the OBI200 and Freeswitch , so you would do
transcoding<br>
with Jitter Buffer and PLC enabled. It would be better if you don't
transcode at all,<br>
but your PSTN provider should offer ILBC or other compressed codec that
you should try and you would also need to rely on them for PLC .You
should probably enable the Jitter Buffer on the bridge if you do
transcoding :<br>
<action application="set"
data="rtp_jitter_buffer_during_bridge=true"/><br>
<br>
Regards,<br>
Dragos<br>
<br>
PS:<br>
Obihai supports Opus in their phones: <a href="http://www.obihai.com/partner/obi1000" target="_blank">http://www.obihai.com/partner/obi1000</a>
.<br>
<br>
<br>
On 11/20/2015 07:22 AM, Russell Treleaven wrote:
<blockquote type="cite">
<div dir="ltr">Are you sure the jitter/loss in on the fixed wireless
side?
<div><br>
</div>
<div>Is the freeswitch<->pstn sip?</div>
<div><br>
</div>
<div>Is freeswitch transcoding between the legs?</div>
<div><br>
</div>
<div>There is a tool called mtr to measure the loss/jitter along the
path.</div>
<div><br>
</div>
<div>If freeswitch is not transcoding then it doesn't need a jitter
buffer.</div>
<div>You could actually use bypass_media to take freeswitch out of
the media path and see if that helps.</div>
<div><a href="https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview</a></div>
<div> </div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Thu, Nov 19, 2015 at 11:07 PM, Josh
Richesin <span dir="ltr"><<a href="javascript:_e(%7B%7D,'cvml','jlrichesin@gmail.com');" target="_blank">jlrichesin@gmail.com</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 dir="ltr">
<div>Hello all!<br>
<br>
</div>
<div>Thanks for the recommendations on upgrading to FS master,
(Assuming that is 1.6.2?) and using the new Jitter Buffer and OPUS
codec. Unfortunately, after researching and working with our ATA
vendor OBI, they do not support OPUS. So that has me form a few more
questions on this issue.<br>
</div>
<div><br>
</div>
<div>1) Is there any documentation on the new Jitter Buffer being
used?<br>
<br>
</div>
<div>2) Is there any reason to believe that upgrading from our
current Free Switch Version 1.4.18 to 1.6.2 would fix the issue with
the current codec of g711, even tho I can not use opus?<br>
<br>
</div>
<div>3) Below is the real issue that I am trying to fix:<br>
</div>
<div><br>
</div>
<div>obi200------------> fixed wireless --------> FreeSwitch
------PSTN</div>
<div><br>
</div>
<div>Call Quality issues are mainly reported by the end user on the
PSTN network. Users report that the calls are cutting in and out but
our users on the obi200 side hear them perfectly. Any ideas on
settings to try to optimize in this situation?<br>
<br>
</div>
<div>I know that it is a network related issue, with high jitter,
however I know there must be a fix as other VOIP providers can ride the
same network path and have no issues at all.<br>
<br>
</div>
<div>4) Main reason for not just upgrading is that I am using
fushionPBX, and they do not yet support 1.6.2, and also trying to
figure out an actual upgrade plan so that I do not loose all the
customers voicemail's, VM greetings passwords and such. <br>
<br>
</div>
<div>Any help is greatly appreciated with what direction I should
go.<br>
<br>
</div>
<div>Thanks!<span><font color="#888888"><br>
</font></span></div>
<span><font color="#888888">
<div><br>
</div>
<div>Josh</div>
</font></span></div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Fri, Nov 13, 2015 at 9:45 AM, Brian
West <span dir="ltr"><<a href="javascript:_e(%7B%7D,'cvml','brian@freeswitch.org');" target="_blank">brian@freeswitch.org</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 dir="ltr">We probably need to wipe confluence clean of the
incorrect jitter buffer docs now that its been completely rewritten in
the past few months.</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Thu, Nov 12, 2015 at 4:06 AM, Ola
Bäckström <span dir="ltr"><<a href="javascript:_e(%7B%7D,'cvml','Ola.Backstrom@loxysoft.se');" target="_blank">Ola.Backstrom@loxysoft.se</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 bgcolor="white" link="blue" vlink="purple" lang="SV">
<div>
<p class="MsoNormal"><span style="font-size:9pt;font-family:"Verdana",sans-serif">Dragos,</span></p>
<p class="MsoNormal"><span style="font-size:9pt;font-family:"Verdana",sans-serif"> </span></p>
<p class="MsoNormal"><span style="font-size:9pt;font-family:"Verdana",sans-serif" lang="EN-US">Your
example to enable jitter buffering uses the letter ”p”. Is that a typo
or does it indeed mean something?
</span></p>
<span>
<p class="MsoNormal"><span lang="EN-US">> You can enable
the Jitter Buffer in the dial plan:<br>
> <action application="set" data="jitterbuffer_msec=2p:25p:"/><br>
<br>
</span><span style="font-size:9pt;font-family:"Verdana",sans-serif" lang="EN-US"></span></p>
</span>
<p class="MsoNormal"><span style="font-size:9pt;font-family:"Verdana",sans-serif" lang="EN-US">I
see no such usage in the documentation at
<a href="https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer</a>
, but just “60:200:20”.
</span></p>
<p class="MsoNormal"><span style="font-size:9pt;font-family:"Verdana",sans-serif" lang="EN-US"> </span></p>
<p class="MsoNormal"><span style="font-size:9pt;font-family:"Verdana",sans-serif" lang="EN-US">Best
regards</span></p>
<p class="MsoNormal"><span style="font-size:9pt;font-family:"Verdana",sans-serif" lang="EN-US">/Ola</span></p>
<p class="MsoNormal"><span style="font-size:9pt;font-family:"Verdana",sans-serif" lang="EN-US"> </span></p>
<div>
<div style="border-style:solid none none;border-color:rgb(225,225,225) -moz-use-text-color -moz-use-text-color;border-width:1pt medium medium;padding:3pt 0cm 0cm">
<p class="MsoNormal"><b><span style="font-size:11pt;font-family:"Calibri",sans-serif;color:windowtext" lang="EN-US">From:</span></b><span style="font-size:11pt;font-family:"Calibri",sans-serif;color:windowtext" lang="EN-US"> <a href="javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>
[mailto:<a href="javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On
Behalf Of </b>Dragos Oancea<br>
<b>Sent:</b> den 12 november 2015 10:14<br>
<b>To:</b> <a href="javascript:_e(%7B%7D,'cvml','freeswitch-users@lists.freeswitch.org');" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b>Subject:</b> Re: [Freeswitch-users] Jitterbuffer</span></p>
</div>
</div>
<div>
<div>
<p class="MsoNormal"><span lang="EN-US"> </span></p>
<p class="MsoNormal">Hi Josh,<br>
<br>
Use FS master, it has a new audio Jitter Buffer.<br>
<br>
Then you can use Opus with this setting in the codec config:<br>
<param name="use-jb-lookahead" value="1"/> . <br>
The other side must be sending FEC so you could profit from it.<br>
<br>
<br>
You can enable the Jitter Buffer in the dial plan:<br>
<action application="set" data="jitterbuffer_msec=2p:25p:"/><br>
And then enable PLC :<br>
<action application="set" data="rtp_jitter_buffer_plc=true"/><br>
<br>
Regards,<br>
Dragos<br>
<br>
<br>
On 11/12/2015 07:51 AM, Josh Richesin wrote: </p>
<blockquote style="margin-top:5pt;margin-bottom:5pt">
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12pt">Hello,</p>
</div>
<p class="MsoNormal" style="margin-bottom:12pt">I am
running FreeSwitch for our small Wireless ISP. Obviously, as a
wireless ISP and ITSP, we have a lot of jitter and some packet loss.
Have a few questions for this setup.</p>
</div>
<p class="MsoNormal" style="margin-bottom:12pt">1) What is
the best codec to use when there is going to be jitter and some
packetloss?</p>
</div>
<p class="MsoNormal" style="margin-bottom:12pt">2) The
ATA's that we are using have adaptive jitter buffers, should I enable
jitterbuffer on Freeswitch as well? If so, should it be via sofia
profiles?</p>
</div>
<p class="MsoNormal" style="margin-bottom:12pt">The problem
that I am trying to solve is this...</p>
</div>
<p class="MsoNormal" style="margin-bottom:12pt">Our voice
customers on our freeswitch have a lot of issues with "calls breaking
up" When they use another voice provider, like say vonage or something
running off of a metaswitch, they do not have the same issues. </p>
</div>
<p class="MsoNormal" style="margin-bottom:12pt">Any help
with this would be appreciated! Thanks in advance.</p>
</div>
<p class="MsoNormal">Josh</p>
</div>
<pre> </pre>
<pre> </pre>
<pre>_________________________________________________________________________</pre>
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<pre> </pre>
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</blockquote>
<p class="MsoNormal"> </p>
</div>
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<br>
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</div>
<br>
<br clear="all">
<div><br>
</div>
-- <br>
<div>
<div dir="ltr">
<div>
<div dir="ltr">
<div>
<div dir="ltr">
<div>
<div dir="ltr">
<div>
<div dir="ltr">
<p><font face="courier new, monospace"><b><i><font size="4">Brian
West</font></i></b><br>
<span style="font-size:x-small"><a href="javascript:_e(%7B%7D,'cvml','brian@freeswitch.org');" target="_blank">brian@freeswitch.org</a></span></font></p>
<p><font face="courier new, monospace" size="1"><img src="http://billing.freeswitch.org/templates/default/img/whmcslogo.png"><br>
</font></p>
<p><font face="monospace, monospace" size="2"><b><i>Twitter:
@FreeSWITCH , @briankwest</i></b><br>
<a href="http://www.freeswitchbook.com" target="_blank">http://www.freeswitchbook.com</a><br>
<a href="http://www.freeswitchcookbook.com" target="_blank">http://www.freeswitchcookbook.com</a></font></p>
<p><font face="monospace, monospace">Got Bugs? Report them <a href="https://freeswitch.org/jira" target="_blank">here</a>! | Reddit: <a href="https://www.reddit.com/r/freeswitch" target="_blank">/r/freeswitch</a></font></p>
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<pre><fieldset></fieldset>
_________________________________________________________________________
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</blockquote>
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</blockquote>