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<div class="moz-cite-prefix">A quick update on this - I have
verified that suppresion of silence suppression(!) on Grandstreams
doesn't work when it is the B-Party of a call. I'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'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 class="moz-txt-link-abbreviated" href="mailto:sofia/internal/0253302@004-0253.sb12.dmclub.org">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 class="moz-txt-link-abbreviated" href="mailto:sofia/internal/0253302@004-0253.sb12.dmclub.org">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 "A" leg of the call (which is placed by a LinkSys SPA922)
I have:<br>
<br>
<sip_local_sdp_str>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</sip_local_sdp_str><br>
<br>
From the troublesome "B" leg of the call (placed by FreeSwitch to
my GXP2000) I have:<br>
<br>
sip_local_sdp_str>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</sip_local_sdp_str><br>
<br>
I was wondering if there's a way I can modify this to include
a="silenceSupp:off - - - -" and whether that would make the
slightest bit of difference to the handset's behaviour!<br>
<br>
<br>
<br>
Alex<br>
</div>
<blockquote cite="mid:512DE07F.3090503@digitalmail.com" type="cite">
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
<div class="moz-cite-prefix">Thanks.<br>
I've been looking at the channel variables of the call. And this
is part of it:<br>
<br>
<rtp_audio_in_raw_bytes>1089091</rtp_audio_in_raw_bytes><rtp_audio_in_media_bytes>957212</rtp_audio_in_media_bytes><rtp_audio_in_packet_count>10344</rtp_audio_in_packet_count><rtp_audio_in_media_packet_count>5581</rtp_audio_in_media_packet_count><rtp_audio_in_skip_packet_count>5480</rtp_audio_in_skip_packet_count><rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count><rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count><rtp_audio_in_cng_packet_count>4323</rtp_audio_in_cng_packet_count><rtp_audio_in_flush_packet_count>440</rtp_audio_in_flush_packet_count><rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size><rtp_audio_out_raw_bytes>1829908</rtp_audio_out_raw_bytes><rtp_audio_out_media_bytes>1829908</rtp_audio_out_media_bytes><rtp_audio_out_packet_count>10639</rtp_audio_out_packet_count><rtp_audio_out
_media_packet_count>10639</rtp_audio_out_media_packet_count><rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count><rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count><rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count><rtp_audio_rtcp_packet_count>0</rtp_audio_rtcp_packet_count><rtp_audio_rtcp_octet_count>0</rtp_audio_rtcp_octet_count><br>
<br>
The thing that slightly concerns me here is
<rtp_audio_in_cng_packet_count>4323</rtp_audio_in_cng_packet_count><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'm afraid it'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
cite="mid:CAKbxfG928nCmqEfGXeeNLA=q3wwpgFs72+7t4WwmxT3zbw+=KA@mail.gmail.com"
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"><<a moz-do-not-send="true"
href="mailto:alex@digitalmail.com" target="_blank">alex@digitalmail.com</a>></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'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>
FreeSWITCH <a moz-do-not-send="true"
href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>
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pstn:+19193869900 </div>
<br>
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<pre wrap="">_________________________________________________________________________
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</pre>
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<p class="" avgcert""="" color="#000000" align="left">No virus
found in this message.<br>
Checked by AVG - <a moz-do-not-send="true"
href="http://www.avg.com">www.avg.com</a><br>
Version: 2012.0.2238 / Virus Database: 2641/5634 - Release
Date: 02/26/13</p>
</blockquote>
<br>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
<a class="moz-txt-link-abbreviated" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
<a class="moz-txt-link-freetext" href="http://www.cudatel.com">http://www.cudatel.com</a>
Official FreeSWITCH Sites
<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.cluecon.com">http://www.cluecon.com</a>
FreeSWITCH-users mailing list
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<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
</pre>
<br>
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<br>
<p class="" avgcert""="" color="#000000" align="left">No virus
found in this message.<br>
Checked by AVG - <a moz-do-not-send="true"
href="http://www.avg.com">www.avg.com</a><br>
Version: 2012.0.2238 / Virus Database: 2641/5635 - Release Date:
02/26/13</p>
</blockquote>
<br>
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