[Freeswitch-users] Silence suppression again

Thu Feb 28 01:20:02 MSK 2013

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)?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>  --
> 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
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2238 / Virus Database: 2641/5634 - Release Date: 02/26/13
>
>
>
>
> _________________________________________________________________________
> 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
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2238 / Virus Database: 2641/5635 - Release Date: 02/26/13
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130227/4143013c/attachment-0001.html 