<div dir="ltr">suppress_cng=true variable or supress-cng profile param set to true.<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Feb 27, 2013 at 11:44 AM, Alex Lake <span dir="ltr">&lt;<a href="mailto:alex@digitalmail.com" target="_blank">alex@digitalmail.com</a>&gt;</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>A quick update on this - I have
      verified that suppresion of silence suppression(!) on Grandstreams
      doesn&#39;t work when it is the B-Party of a call. I&#39;m not sure if
      silence-suppression is part of the SIP handshake, but I know that
      other VOIP technologies seem to be able to prevent it.<br>
      I&#39;m calling in a handset specialist to advise..<br>
      <br>
      One of the things that intrigues me is this extract from the log
      (particularly the reference to silenceSupp):<br>
      <br>
      2013-02-27 17:28:05.151624 [DEBUG] switch_core_codec.c:244
      <a href="mailto:sofia/internal/0253302@004-0253.sb12.dmclub.org" target="_blank">sofia/internal/0253302@004-0253.sb12.dmclub.org</a> Restore previous
      codec PCMA:8.<br>
      2013-02-27 17:28:05.151624 [DEBUG] mod_sofia.c:754 Local SDP
      <a href="mailto:sofia/internal/0253302@004-0253.sb12.dmclub.org" target="_blank">sofia/internal/0253302@004-0253.sb12.dmclub.org</a>:<br>
      v=0<br>
      o=FreeSWITCH 1361975871 1361975873 IN IP4 176.58.88.201<br>
      s=FreeSWITCH<br>
      c=IN IP4 176.58.88.201<br>
      t=0 0<br>
      m=audio 10212 RTP/AVP 8 101<br>
      a=rtpmap:8 PCMA/8000<br>
      a=rtpmap:101 telephone-event/8000<br>
      a=fmtp:101 0-16<br>
      a=silenceSupp:off - - - -<br>
      a=ptime:30<br>
      a=sendrecv<br>
      <br>
      What does all this mean?<br>
      <br>
      From the &quot;A&quot; leg of the call (which is placed by a LinkSys SPA922)
      I have:<br>
      <br>
&lt;sip_local_sdp_str&gt;v%3D0%0Ao%3DFreeSWITCH%201361974765%201361974767%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011468%20RTP/AVP%208%20101%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0A&lt;/sip_local_sdp_str&gt;<br>

      <br>
      From the troublesome &quot;B&quot; leg of the call (placed by FreeSwitch to
      my GXP2000) I have:<br>
      <br>
sip_local_sdp_str&gt;v%3D0%0Ao%3DFreeSWITCH%201361974745%201361974746%20IN%20IP4%20176.58.88.201%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%20176.58.88.201%0At%3D0%200%0Am%3Daudio%2011488%20RTP/AVP%208%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A30%0Aa%3Dsendrecv%0Am%3Daudio%2011488%20RTP/AVP%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A&lt;/sip_local_sdp_str&gt;<br>

      <br>
      I was wondering if there&#39;s a way I can modify this to include
      a=&quot;silenceSupp:off - - - -&quot; and whether that would make the
      slightest bit of difference to the handset&#39;s behaviour!<br>
      <br>
      <br>
      <br>
      Alex<br>
    </div>
    <blockquote type="cite"><div><div class="h5">
      
      <div>Thanks.<br>
        I&#39;ve been looking at the channel variables of the call. And this
        is part of it:<br>
        <br>
&lt;rtp_audio_in_raw_bytes&gt;1089091&lt;/rtp_audio_in_raw_bytes&gt;&lt;rtp_audio_in_media_bytes&gt;957212&lt;/rtp_audio_in_media_bytes&gt;&lt;rtp_audio_in_packet_count&gt;10344&lt;/rtp_audio_in_packet_count&gt;&lt;rtp_audio_in_media_packet_count&gt;5581&lt;/rtp_audio_in_media_packet_count&gt;&lt;rtp_audio_in_skip_packet_count&gt;5480&lt;/rtp_audio_in_skip_packet_count&gt;&lt;rtp_audio_in_jb_packet_count&gt;0&lt;/rtp_audio_in_jb_packet_count&gt;&lt;rtp_audio_in_dtmf_packet_count&gt;0&lt;/rtp_audio_in_dtmf_packet_count&gt;&lt;rtp_audio_in_cng_packet_count&gt;4323&lt;/rtp_audio_in_cng_packet_count&gt;&lt;rtp_audio_in_flush_packet_count&gt;440&lt;/rtp_audio_in_flush_packet_count&gt;&lt;rtp_audio_in_largest_jb_size&gt;0&lt;/rtp_audio_in_largest_jb_size&gt;&lt;rtp_audio_out_raw_bytes&gt;1829908&lt;/rtp_audio_out_raw_bytes&gt;&lt;rtp_audio_out_media_bytes&gt;1829908&lt;/rtp_audio_out_media_bytes&gt;&lt;rtp_audio_out_packet_count&gt;10639&lt;/rtp_audio_out_packet_count&gt;&lt;rtp_audio_out

_media_packet_count&gt;10639&lt;/rtp_audio_out_media_packet_count&gt;&lt;rtp_audio_out_skip_packet_count&gt;0&lt;/rtp_audio_out_skip_packet_count&gt;&lt;rtp_audio_out_dtmf_packet_count&gt;0&lt;/rtp_audio_out_dtmf_packet_count&gt;&lt;rtp_audio_out_cng_packet_count&gt;0&lt;/rtp_audio_out_cng_packet_count&gt;&lt;rtp_audio_rtcp_packet_count&gt;0&lt;/rtp_audio_rtcp_packet_count&gt;&lt;rtp_audio_rtcp_octet_count&gt;0&lt;/rtp_audio_rtcp_octet_count&gt;<br>

        <br>
        The thing that slightly concerns me here is
&lt;rtp_audio_in_cng_packet_count&gt;4323&lt;/rtp_audio_in_cng_packet_count&gt;<br>
        <br>
        That sounds as though the inbound audio path contains some
        comfort noise, suggesting that the handset the other end has
        some kind of silence suppression enabled.<br>
        <br>
        I&#39;m afraid it&#39;s a dreaded Grandstream GXP2000 - which I know are
        not the handset of the cognoscenti! - but the customer tells me
        that he has silence suppression disabled. Do we think this is
        correct?!<br>
        <br>
        Rgds,<br>
        Alex<br>
      </div>
      <blockquote type="cite">
        <div dir="ltr">It does not send any by default.  Also, yes it
          only generates silence when the call is not getting any RTP.
          <div><br>
          </div>
        </div>
        <div class="gmail_extra"><br>
          <br>
          <div class="gmail_quote">On Tue, Feb 26, 2013 at 12:00 PM,
            Alex Lake <span dir="ltr">&lt;<a href="mailto:alex@digitalmail.com" target="_blank">alex@digitalmail.com</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Does
              Freeswitch enable any kind of silence suppression by
              default? If I<br>
              have bridge_generate_comfort_noise=true, does that only
              have an effect<br>
              when the audio-generating end decides to stop sending RTP?<br>
              <br>
              Any other tips for diagnosing silence (it&#39;s kind of like a
              slow noise<br>
              gate effect)?<br>
              <br>
_________________________________________________________________________<br>
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            </blockquote>
          </div>
          <br>
          <br clear="all">
          <div><br>
          </div>
          -- <br>
          Anthony Minessale II<br>
          <br>
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          pstn:<a href="tel:%2B19193869900" value="+19193869900" target="_blank">+19193869900</a> </div>
        <br>
        <fieldset></fieldset>
        <br>
        <pre>_________________________________________________________________________
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</pre>
        <br>
        <fieldset></fieldset>
        <br>
        <p color="#000000" align="left">No virus
          found in this message.<br>
          Checked by AVG - <a href="http://www.avg.com" target="_blank">www.avg.com</a><br>
          Version: 2012.0.2238 / Virus Database: 2641/5634 - Release
          Date: 02/26/13</p>
      </blockquote>
      <br>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_________________________________________________________________________
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</pre>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><p color="#000000" align="left"></p><div><div class="h5">No virus
        found in this message.<br>
        Checked by AVG - <a href="http://www.avg.com" target="_blank">www.avg.com</a><br></div></div>
        Version: 2012.0.2238 / Virus Database: 2641/5635 - Release Date:
        02/26/13<p></p>
    </blockquote>
    <br>
  </div>

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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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pstn:+19193869900
</div>