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

Fri Apr 11 21:18:49 MSD 2014

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 30i on 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
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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
> 
> 
> 
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> ______________________________________________________________________
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> 
>  
> 
> 
> Official FreeSWITCH Sites
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