[Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file

Fri Apr 11 22:11:18 MSD 2014

This link should be informative

http://www.opus-codec.org/comparison/

G722 uses the same amount of bandwidth as PCMU/PCMA.

On Fri, Apr 11, 2014 at 1:35 PM, Moishe Grunstein <max at nysolutions.com>wrote:

> G711A/U is probably the most compatible or used codec, If you lock a
> system and all hardware/software to only use G711 you won't you can avoid
> transcoding. G722 is a HD codec, uses more bandwidth and some carriers or
> hardware may not support it and you may be forced to transcode. If you have
> limited bandwidth you may be forced to use a low bandwidth codec which may
> force you to transcode.
>
>
> Thanks,
>
> Moishe Grunstein
> Tornado Computer Systems, Inc.
> 212.400.7650 888.IPPBX.US
> Service Request Email: support at nysolutions.com
> Polycom Certified VAR
> Microsoft Small Business Specialist, Cisco SMB Select Certified
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy
> Sent: Friday, April 11, 2014 1:19 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific
> MOH file
>
> HI All,
>
> It appears that the user was helping more than I thought.  I had specified
> pref codec G722 and second and third as "unspecified".  Then there are a
> series of dropdown list boxes for lots of codecs to be enabled yes or no.
>
> The user had changed G722 to enabled=NO.  This confused me why the
> preferred was not being used.
>
> After re-enabling G722, the problem is resolved.
>
> But this left me with CODEC questions and the wiki has not answered them
> for me.  I had read that G722 was the "HD" codec for improved voice
> quality.  I have also read elsewhere that PCMU or PCMA will give the best
> quality at the expense of bandwidth.  Now it appears that G722 uses the
> same bandwidth as PCMA/U.   Very confusing.
>
> So, if bandwidth is not a concern (at this time), what codec is going to
> give the best sound quality?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Friday, April 11, 2014 12:27 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific
> MOH file
>
> Looks like its changing on hold/unhold, set it to 0.030, and do PCMU at 30ion your profile or for that user.
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9001
> ISN: 410*543
> Skype:briankwest
> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Apr 11, 2014, at 11:19 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
> > HI Moishe,
> >
> > Thanks for responding.  I do not have a wireshark of this incident.  I
> will see about getting one.  I was not sure if it would show anything
> relevant.
> >
> > With regard to transcoding, I did not think so.  I am reviewing the logs
> in pastebin again.
> >
> > The RTP packet size on all phones is 0.020.  FS is 0.030, but
> auto-adjusts to 0.020.
> >
> > I do see at about line 317 in pastebin that the transfer connection
> appears to be going to PCMU and rtp30:
> >
> > ...c2f v=0
> > ...c2f o=- 12482737 12482737 IN IP4 71.179.239.241 ...c2f s=- ...c2f
> > c=IN IP4 71.179.239.241 ...c2f t=0 0 ...c2f m=audio 16490 RTP/AVP 0
> > 101 ...c2f a=rtpmap:0 PCMU/8000 ...c2f a=rtpmap:101
> > telephone-event/8000 ...c2f a=fmtp:101 0-15 ...c2f a=ptime:30 ...c2f
> > 2014-04-01 12:01:51.989154 [DEBUG] switch_core_session.c:1016 Send
> > signal sofia/external/sip:203 at 71.179.239.241:1467 [BREAK] ...c2f
> > 2014-04-01 12:01:51.989154 [DEBUG] switch_core_session.c:1016 Send
> > signal sofia/external/sip:203 at 71.179.239.241:1467 [BREAK] ...c2f
> > 2014-04-01 12:01:51.989154 [DEBUG] sofia.c:5815 Channel
> > sofia/external/sip:203 at 71.179.239.241:1467 entering state [ready][200]
> > ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5282 Audio
> > Codec Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000]
> > ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5282 Audio
> > Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000]
> > ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5282 Audio
> > Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000]
> > ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5282 Audio
> > Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200]
> > ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5355
> > Substituting codec PCMU at 30i@8000h ...c2f 2014-04-01 12:01:51.989154
> > [DEBUG] sofia_glue.c:3190 Set Codec
> > sofia/external/sip:203 at 71.179.239.241:1467 PCMU/8000 30 ms 240 samples
> > 64000 bits ...c2f 2014-04-01 12:01:51.989154 [DEBUG]
> > switch_core_codec.c:111 sofia/external/sip:203 at 71.179.239.241:1467
> > Original read codec set to PCMU:0
> >
> > I know I checked and they are set to G722 and use primary only.
> >
> > I will check again and get a new log dump.
> >
> > Sean
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Moishe Grunstein
> > Sent: Thursday, April 10, 2014 11:44 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with
> > specific MOH file
> >
> > Do you have a wireshark? Any transcoding going on? What do you have the
> rtp packet size set to on all phones?
> >
> >
> > Thanks,
> >
> > Moishe Grunstein
> > Tornado Computer Systems, Inc.
> > 212.400.7650 888.IPPBX.US
> > Service Request Email: support at nysolutions.com Polycom Certified VAR
> > Microsoft Small Business Specialist, Cisco SMB Select Certified
> >
> > Computer Networking * Managed Services * IP Video Surveillance *
> > Network Assessments * Web Solutions * Voice over IP * Disaster
> > Recovery * Network Security * Site Surveys * CMS
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Sean Devoy
> > Sent: Thursday, April 10, 2014 10:46 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with
> > specific MOH file
> >
> > Hi Everyone,
> >
> > I have hit a dead end on this.  I would say she was just crazy except I
> can reproduce the problem every time.
> >
> > I have checked her phone setup through the web interface and I cannot
> find any field that is different from a working phone (except user id and
> password).  The Directory entries match as well.
> >
> > Please, any suggestions on further testing/logging I could do would be
> most welcome.
> >
> > I am ready to put a $50 bounty on the first person to provide a
> resolution.  If it is unclear I will do my best to split it up!  This
> customer is very important and now is very frustrated.
> >
> > To recap for anyone who does not recognize the thread:
> > When my user at 203 answers incoming calls or places outgoing calls,
> everything works perfectly.  However, if she does not answer and someone
> else picks up (i.e. her extension gets a LOSE RACE result) and then
> transfers to her the audio on the CALLER's end is "muffled".  She says
> several people have asked if she was under water.  I thought it sounded
> like someone speaking too close to a microphone.  All phones are CISCO
> SPA504G models.
> >
> > There is anecdotal evidence of delayed audio, not delayed connection,
> but ongoing delayed speech.
> >
> > This ONLY happens when a call in XFERed to her and it happens every time
> that occurs. It does not occur if she answers and puts a call on hold and
> picks it back up.
> >
> > I could test having her xfer one of the bad audio calls back to someone
> else.  But, I must say I don't know if it would tell me anything either
> way!  I suppose I could swap out her physical phone, but that seems HIGHLY
> unlikely.
> >
> > The log file is in pastebin:
> > http://pastebin.freeswitch.org/22302
> >
> > Thanks,
> > Sean
> >
> >
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > 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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> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
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> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
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> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
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> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
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>
>
> _________________________________________________________________________
> 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
>
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>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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> http://www.cluecon.com
>
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