suppress_cng=true variable or supress-cng profile param set to true.


On Wed, Feb 27, 2013 at 11:44 AM, Alex Lake <alex@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@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@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@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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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http://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@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://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@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@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@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900