[Freeswitch-users] Silence suppression again

Thu Feb 28 11:57:49 MSK 2013

Ah! I think that has fixed it. Many thanks. I'll see if I can update a 
bit of wiki somewhere...
> suppress_cng=true variable or supress-cng profile param set to true.
>
>
> On Wed, Feb 27, 2013 at 11:44 AM, Alex Lake <alex at digitalmail.com 
> <mailto:alex at digitalmail.com>> wrote:
>
>     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.
>     I'm calling in a handset specialist to advise..
>
>     One of the things that intrigues me is this extract from the log
>     (particularly the reference to silenceSupp):
>
>     2013-02-27 17:28:05.151624 [DEBUG] switch_core_codec.c:244
>     sofia/internal/0253302 at 004-0253.sb12.dmclub.org
>     <mailto:sofia/internal/0253302 at 004-0253.sb12.dmclub.org> Restore
>     previous codec PCMA:8.
>     2013-02-27 17:28:05.151624 [DEBUG] mod_sofia.c:754 Local SDP
>     sofia/internal/0253302 at 004-0253.sb12.dmclub.org
>     <mailto:sofia/internal/0253302 at 004-0253.sb12.dmclub.org>:
>     v=0
>     o=FreeSWITCH 1361975871 1361975873 IN IP4 176.58.88.201
>     s=FreeSWITCH
>     c=IN IP4 176.58.88.201
>     t=0 0
>     m=audio 10212 RTP/AVP 8 101
>     a=rtpmap:8 PCMA/8000
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-16
>     a=silenceSupp:off - - - -
>     a=ptime:30
>     a=sendrecv
>
>     What does all this mean?
>
>     From the "A" leg of the call (which is placed by a LinkSys SPA922)
>     I have:
>
>     <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>
>
>     From the troublesome "B" leg of the call (placed by FreeSwitch to
>     my GXP2000) I have:
>
>     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>
>
>     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!
>
>
>
>     Alex
>>     Thanks.
>>     I've been looking at the channel variables of the call. And this
>>     is part of it:
>>
>>     <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>
>>
>>     The thing that slightly concerns me here is
>>     <rtp_audio_in_cng_packet_count>4323</rtp_audio_in_cng_packet_count>
>>
>>     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.
>>
>>     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?!
>>
>>     Rgds,
>>     Alex
>>>     It does not send any by default.  Also, yes it only generates
>>>     silence when the call is not getting any RTP.
>>>
>>>
>>>
>>>     On Tue, Feb 26, 2013 at 12:00 PM, Alex Lake
>>>     <alex at digitalmail.com <mailto:alex at digitalmail.com>> wrote:
>>>
>>>         Does Freeswitch enable any kind of silence suppression by
>>>         default? If I
>>>         have bridge_generate_comfort_noise=true, does that only have
>>>         an effect
>>>         when the audio-generating end decides to stop sending RTP?
>>>
>>>         Any other tips for diagnosing silence (it's kind of like a
>>>         slow noise
>>>         gate effect)?
>>>
>>>         _________________________________________________________________________
>>>         Professional FreeSWITCH Consulting Services:
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>>>         http://www.freeswitchsolutions.com
>>>
>>>         
>>>         
>>>
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>>>
>>>
>>>
>>>     -- 
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>     Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>>     AIM: anthm
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>>>
>>>
>>>     _________________________________________________________________________
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>>>     http://www.freeswitchsolutions.com
>>>
>>>     
>>>     
>>>
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>>>
>>>     No virus found in this message.
>>>     Checked by AVG - www.avg.com <http://www.avg.com>
>>>     Version: 2012.0.2238 / Virus Database: 2641/5634 - Release Date:
>>>     02/26/13
>>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
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>>
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>>
>>     No virus found in this message.
>>     Checked by AVG - www.avg.com <http://www.avg.com>
>>     Version: 2012.0.2238 / Virus Database: 2641/5635 - Release Date:
>>     02/26/13 
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
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>
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>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
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> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2238 / Virus Database: 2641/5636 - Release Date: 02/27/13
>

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