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<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A">Thanks Russell and Moishe,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A">After I sent that and thought it through more thoroughly it occurred to me the answer is “The best codec is the one that avoids transcoding and has the highest audio quality
on the other end.”<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A">For the record, the resolution was to be sure G722, PCMA and PCMU were all enabled on the device and in my case select G722 as preferred but allow others.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A">Moishe has declined the bounty in favor of a donation the FS Dev’s in his name. I will send it shortly.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A">Sean<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Verdana","sans-serif";color:#44546A"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
<b>On Behalf Of </b>Russell Treleaven<br>
<b>Sent:</b> Friday, April 11, 2014 2:11 PM<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<div>
<div>
<p class="MsoNormal">This link should be informative<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal"><a href="http://www.opus-codec.org/comparison/">http://www.opus-codec.org/comparison/</a><o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">G722 uses the same amount of bandwidth as PCMU/PCMA.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
<div>
<p class="MsoNormal">On Fri, Apr 11, 2014 at 1:35 PM, Moishe Grunstein <<a href="mailto:max@nysolutions.com" target="_blank">max@nysolutions.com</a>> wrote:<o:p></o:p></p>
<blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class="MsoNormal">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.<o:p></o:p></p>
<div>
<p class="MsoNormal"><br>
<br>
Thanks,<br>
<br>
Moishe Grunstein<br>
Tornado Computer Systems, Inc.<br>
212.400.7650 <a href="http://888.IPPBX.US" target="_blank">888.IPPBX.US</a><br>
Service Request Email: <a href="mailto:support@nysolutions.com">support@nysolutions.com</a><br>
Polycom Certified VAR<br>
Microsoft Small Business Specialist, Cisco SMB Select Certified<br>
<br>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS<br>
<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Sean
Devoy<o:p></o:p></p>
</div>
<div>
<div>
<p class="MsoNormal">Sent: Friday, April 11, 2014 1:19 PM<br>
To: FreeSWITCH Users Help<br>
Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file<br>
<br>
HI All,<br>
<br>
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.<br>
<br>
The user had changed G722 to enabled=NO. This confused me why the preferred was not being used.<br>
<br>
After re-enabling G722, the problem is resolved.<br>
<br>
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.<br>
<br>
So, if bandwidth is not a concern (at this time), what codec is going to give the best sound quality?<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Brian
West<br>
Sent: Friday, April 11, 2014 12:27 PM<br>
To: FreeSWITCH Users Help<br>
Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file<br>
<br>
Looks like its changing on hold/unhold, set it to 0.030, and do PCMU@30i on your profile or for that user.<br>
--<br>
Brian West<br>
<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a><br>
FreeSWITCH Solutions, LLC<br>
PO BOX 2531<br>
Brookfield, WI 53008-2531<br>
Twitter: @FreeSWITCH , @briankwest<br>
<a href="http://www.freeswitchbook.com" target="_blank">http://www.freeswitchbook.com</a><br>
<a href="http://www.freeswitchcookbook.com" target="_blank">http://www.freeswitchcookbook.com</a><br>
<br>
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST<br>
iNUM: +883 5100 1420 9001<br>
ISN: 410*543<br>
Skype:briankwest<br>
PGP Key: <a href="http://www.bkw.org/key.txt" target="_blank">http://www.bkw.org/key.txt</a> (AB93356707C76CED)<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
On Apr 11, 2014, at 11:19 AM, Sean Devoy <<a href="mailto:sdevoy@bizfocused.com">sdevoy@bizfocused.com</a>> wrote:<br>
<br>
> HI Moishe,<br>
><br>
> 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.<br>
><br>
> With regard to transcoding, I did not think so. I am reviewing the logs in pastebin again.<br>
><br>
> The RTP packet size on all phones is 0.020. FS is 0.030, but auto-adjusts to 0.020.<br>
><br>
> I do see at about line 317 in pastebin that the transfer connection appears to be going to PCMU and rtp30:<br>
><br>
> ...c2f v=0<br>
> ...c2f o=- 12482737 12482737 IN IP4 71.179.239.241 ...c2f s=- ...c2f<br>
> c=IN IP4 71.179.239.241 ...c2f t=0 0 ...c2f m=audio 16490 RTP/AVP 0<br>
> 101 ...c2f a=rtpmap:0 PCMU/8000 ...c2f a=rtpmap:101<br>
> telephone-event/8000 ...c2f a=fmtp:101 0-15 ...c2f a=ptime:30 ...c2f<br>
> 2014-04-01 12:01:51.989154 [DEBUG] switch_core_session.c:1016 Send<br>
> signal sofia/external/<a href="http://sip:203@71.179.239.241:1467" target="_blank">sip:203@71.179.239.241:1467</a> [BREAK] ...c2f<br>
> 2014-04-01 12:01:51.989154 [DEBUG] switch_core_session.c:1016 Send<br>
> signal sofia/external/<a href="http://sip:203@71.179.239.241:1467" target="_blank">sip:203@71.179.239.241:1467</a> [BREAK] ...c2f<br>
> 2014-04-01 12:01:51.989154 [DEBUG] sofia.c:5815 Channel<br>
> sofia/external/<a href="http://sip:203@71.179.239.241:1467" target="_blank">sip:203@71.179.239.241:1467</a> entering state [ready][200]<br>
> ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5282 Audio<br>
> Codec Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000]<br>
> ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5282 Audio<br>
> Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000]<br>
> ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5282 Audio<br>
> Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000]<br>
> ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5282 Audio<br>
> Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200]<br>
> ...c2f 2014-04-01 12:01:51.989154 [DEBUG] sofia_glue.c:5355<br>
> Substituting codec PCMU@30i@8000h ...c2f 2014-04-01 12:01:51.989154<br>
> [DEBUG] sofia_glue.c:3190 Set Codec<br>
> sofia/external/<a href="http://sip:203@71.179.239.241:1467" target="_blank">sip:203@71.179.239.241:1467</a> PCMU/8000 30 ms 240 samples<br>
> 64000 bits ...c2f 2014-04-01 12:01:51.989154 [DEBUG]<br>
> switch_core_codec.c:111 sofia/external/<a href="http://sip:203@71.179.239.241:1467" target="_blank">sip:203@71.179.239.241:1467</a><br>
> Original read codec set to PCMU:0<br>
><br>
> I know I checked and they are set to G722 and use primary only.<br>
><br>
> I will check again and get a new log dump.<br>
><br>
> Sean<br>
> -----Original Message-----<br>
> From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of<br>
> Moishe Grunstein<br>
> Sent: Thursday, April 10, 2014 11:44 PM<br>
> To: FreeSWITCH Users Help<br>
> Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with<br>
> specific MOH file<br>
><br>
> Do you have a wireshark? Any transcoding going on? What do you have the rtp packet size set to on all phones?<br>
><br>
><br>
> Thanks,<br>
><br>
> Moishe Grunstein<br>
> Tornado Computer Systems, Inc.<br>
> 212.400.7650 <a href="http://888.IPPBX.US" target="_blank">888.IPPBX.US</a><br>
> Service Request Email: <a href="mailto:support@nysolutions.com">support@nysolutions.com</a> Polycom Certified VAR<br>
> Microsoft Small Business Specialist, Cisco SMB Select Certified<br>
><br>
> Computer Networking * Managed Services * IP Video Surveillance *<br>
> Network Assessments * Web Solutions * Voice over IP * Disaster<br>
> Recovery * Network Security * Site Surveys * CMS<br>
><br>
> -----Original Message-----<br>
> From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of<br>
> Sean Devoy<br>
> Sent: Thursday, April 10, 2014 10:46 PM<br>
> To: FreeSWITCH Users Help<br>
> Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with<br>
> specific MOH file<br>
><br>
> Hi Everyone,<br>
><br>
> I have hit a dead end on this. I would say she was just crazy except I can reproduce the problem every time.<br>
><br>
> 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.<br>
><br>
> Please, any suggestions on further testing/logging I could do would be most welcome.<br>
><br>
> 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.<br>
><br>
> To recap for anyone who does not recognize the thread:<br>
> 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.<br>
><br>
> There is anecdotal evidence of delayed audio, not delayed connection, but ongoing delayed speech.<br>
><br>
> 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.<br>
><br>
> 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.<br>
><br>
> The log file is in pastebin:<br>
> <a href="http://pastebin.freeswitch.org/22302" target="_blank">http://pastebin.freeswitch.org/22302</a><br>
><br>
> Thanks,<br>
> Sean<br>
><br>
><br>
><br>
> ______________________________________________________________________<br>
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<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><o:p></o:p></p>
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