<div dir="ltr"><div>This link should be informative</div><div><br></div><div><a href="http://www.opus-codec.org/comparison/">http://www.opus-codec.org/comparison/</a><br></div><div><br></div><div>G722 uses the same amount of bandwidth as PCMU/PCMA.<br>
</div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Apr 11, 2014 at 1:35 PM, Moishe Grunstein <span dir="ltr"><<a href="mailto:max@nysolutions.com" target="_blank">max@nysolutions.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">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.<br>
<div class="im HOEnZb"><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<br>
</div><div class="HOEnZb"><div class="h5">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>
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<br>
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<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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</div></div></blockquote></div><br></div>