Alex,<br><br>I suspect these pages would be a good place to start:<br><a href="http://wiki.freeswitch.org/wiki/VAD_and_CNG">http://wiki.freeswitch.org/wiki/VAD_and_CNG</a><br><a href="http://wiki.freeswitch.org/wiki/Variable_suppress_cng">http://wiki.freeswitch.org/wiki/Variable_suppress_cng</a><br>
<br>-MC<br><br><div class="gmail_quote">On Thu, Feb 28, 2013 at 12:57 AM, Alex Lake <span dir="ltr"><<a 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">
<div text="#000000" bgcolor="#FFFFFF">
<div>Ah! I think that has fixed it. Many
thanks. I'll see if I can update a bit of wiki somewhere...<br>
</div>
<blockquote type="cite">
<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"><<a 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">
<div text="#000000" bgcolor="#FFFFFF">
<div>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 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 "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 type="cite">
<div>
<div>
<div>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 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 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>
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<p color="#000000" align="left">No virus found in
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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>
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<div>No virus found in this message.<br>
Checked by AVG - <a href="http://www.avg.com" target="_blank">www.avg.com</a><br>
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Version: 2012.0.2238 / Virus Database: 2641/5635 -
Release Date: 02/26/13 </blockquote>
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-- <br>
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<br>
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<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: 2641/5636 - Release Date:
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</blockquote>
<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
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