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

Fri Apr 11 20:19:32 MSD 2014

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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