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    <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>
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</style>Regards,<br>
      Bipin<br>
      <br>
      <br>
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    <div class="moz-cite-prefix">-------- Original Message --------<br>
      Subject: [Freeswitch-users] Robotic voice with g729 pass through<br>
      From: devang nathwani <a class="moz-txt-link-rfc2396E" href="mailto:devang.nathwani31589@gmail.com">&lt;devang.nathwani31589@gmail.com&gt;</a><br>
      To: FreeSWITCH Users Help
      <a class="moz-txt-link-rfc2396E" href="mailto:freeswitch-users@lists.freeswitch.org">&lt;freeswitch-users@lists.freeswitch.org&gt;</a><br>
      Date: 9/1/2016, 11:43:38 AM<br>
    </div>
    <blockquote
cite="mid:CAGOm0PL7hj8bmAzq93DPBTFJCEL1eFPDsF+mezLkOY0KXKbvcw@mail.gmail.com"
      type="cite">
      <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
          moz-do-not-send="true" href="http://pastebin.com/D2cPLJWe">http://pastebin.com/D2cPLJWe</a></div>
      <br>
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      <br>
      <pre wrap="">_________________________________________________________________________
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