<div dir="ltr"><div>Hello Bipin,<br><br></div>Thanks for your suggestion and time, we have applied all there params respectively in dialplan as well as added in sip profile. we have play around with all there by disable one and enabling other but still i am getting the choppy, robotic sound. I can say bit difference in audio result in good way but nothing remarkable.<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Sep 1, 2016 at 2:19 PM, Bipin Patel <span dir="ltr"><<a href="mailto:bipin@xbipin.com" target="_blank">bipin@xbipin.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<font size="-1"><font face="Arial">hi,<br>
<br>
do u have jitter buffer enabled because i recall i had a similar
issue and after disabling jitter buffer, cng to false and vad to
none it resolved<br>
<br>
</font></font>
<div>
Regards,<br>
Bipin<br>
<br>
<br>
<hr>
</div><div><div class="h5">
<div>-------- Original Message --------<br>
Subject: [Freeswitch-users] Robotic voice with g729 pass through<br>
From: devang nathwani <a href="mailto:devang.nathwani31589@gmail.com" target="_blank"><devang.nathwani31589@gmail.<wbr>com></a><br>
To: FreeSWITCH Users Help
<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank"><freeswitch-users@lists.<wbr>freeswitch.org></a><br>
Date: 9/1/2016, 11:43:38 AM<br>
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr">Hello All,<br>
<br>
Whenever i am trying to make a call, Two way commnunication was
established but I am having kind of choppy like robotic voice or
very bad quality of voice with only G729 codec.For PCMU,PCMA
both are working fine.I want to work with G729.<br>
I have set below parameters in sip profile to try to resolve
this issue. But it doesn't work for me.<br>
<br>
aggressive-nat-detection : true<br>
absolute_codec_string : G729<br>
inbound-codec-prefs :G729,PCMU,PCMA<br>
outbound-codec-prefs : G729,PCMU,PCMA<br>
inbound-late-negotiation :true<br>
inbound-codec-negotiation : generous<br>
<br>
Any help and ideas will be appreciated.<br>
<br>
FYI : For debug purpose you may get SIP Log from <a href="http://pastebin.com/D2cPLJWe" target="_blank">http://pastebin.com/D2cPLJWe</a></div>
<br>
<fieldset></fieldset>
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