[Freeswitch-users] Silence suppression again

Fri Mar 1 03:54:58 MSK 2013

Alex,

I suspect these pages would be a good place to start:
http://wiki.freeswitch.org/wiki/VAD_and_CNG
http://wiki.freeswitch.org/wiki/Variable_suppress_cng

-MC

On Thu, Feb 28, 2013 at 12:57 AM, Alex Lake <alex at digitalmail.com> wrote:

>  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> 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 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:
>> 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> 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)?
>>>
>>> _________________________________________________________________________
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>>>
>>> 
>>> 
>>>
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>>
>>
>>
>>  --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
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>>
>>
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>>
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>>
>>
>>
>>
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>>
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>>
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>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
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>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2238 / Virus Database: 2641/5636 - Release Date: 02/27/13
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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-- 
Michael S Collins
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