Looks like its changing on hold/unhold, set it to 0.030, and do PCMU@30i on your profile or for that user. -- Brian West brian@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@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@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@71.179.239.241:1467 [BREAK] ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia.c:5815 Channel sofia/external/sip:203@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@30i@8000h ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:3190 Set Codec sofia/external/sip:203@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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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