Audio degrades ON TRANSFER with specific MOH file
Hi All, I have a user that has claimed that if she picks up incoming calls, the audio is excellent. However, if someone else picks up the call and transfers it to her, the audio is "terrible". While investigating I found that they are using their own custom MOH file. I suspect that the AUDIO configuration is being modified by the MOH feature and leaving her transferred call in a degraded state. But, I don't understand enough to actually see if that is the case. Please review the pastebin file: http://pastebin.freeswitch.org/22302 I think I have trimmed it down to the relevant data. Ext 302 has answered and is just initiating the transfer. If I am correct in my hypothesis, what is the correct solution? Thanks in advance, Sean
You need to enable rewrite-rtp-timestamps-option on the profiles. Probably after a few seconds, ... the audio will stabilise again (can also depend on endpoints). Anyway, what you're probably experiencing is that after the transfer the rtp-timestamps aren't correct anymore. Kind regards, Michel On Thu, Apr 3, 2014 at 6:05 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi All,
I have a user that has claimed that if she picks up incoming calls, the audio is excellent. However, if someone else picks up the call and transfers it to her, the audio is "terrible".
While investigating I found that they are using their own custom MOH file. I suspect that the AUDIO configuration is being modified by the MOH feature and leaving her transferred call in a degraded state. But, I don't understand enough to actually see if that is the case.
Please review the pastebin file: http://pastebin.freeswitch.org/22302
I think I have trimmed it down to the relevant data. Ext 302 has answered and is just initiating the transfer.
If I am correct in my hypothesis, what is the correct solution?
Thanks in advance,
Sean
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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I would venture to guess that its an issue with G722 to PCMU hold/unhold. Just glancing over the logs, What phone are you using? -- 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 3, 2014, at 7:44 AM, Michel Brabants <michel.brabants@gmail.com> wrote:
You need to enable rewrite-rtp-timestamps-option on the profiles. Probably after a few seconds, ... the audio will stabilise again (can also depend on endpoints). Anyway, what you're probably experiencing is that after the transfer the rtp-timestamps aren't correct anymore.
Kind regards,
Michel
On Thu, Apr 3, 2014 at 6:05 AM, Sean Devoy <sdevoy@bizfocused.com> wrote: <image001.gif> Hi All,
I have a user that has claimed that if she picks up incoming calls, the audio is excellent. However, if someone else picks up the call and transfers it to her, the audio is “terrible”.
While investigating I found that they are using their own custom MOH file. I suspect that the AUDIO configuration is being modified by the MOH feature and leaving her transferred call in a degraded state. But, I don’t understand enough to actually see if that is the case.
Please review the pastebin file: http://pastebin.freeswitch.org/22302
I think I have trimmed it down to the relevant data. Ext 302 has answered and is just initiating the transfer.
If I am correct in my hypothesis, what is the correct solution?
Thanks in advance,
Sean
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hey Brian, All Cisco 504G phones. Sean -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, April 03, 2014 1:13 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file I would venture to guess that its an issue with G722 to PCMU hold/unhold. Just glancing over the logs, What phone are you using? -- 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 3, 2014, at 7:44 AM, Michel Brabants <michel.brabants@gmail.com> wrote:
You need to enable rewrite-rtp-timestamps-option on the profiles. Probably after a few seconds, ... the audio will stabilise again (can also depend on endpoints). Anyway, what you're probably experiencing is that after the transfer the rtp-timestamps aren't correct anymore.
Kind regards,
Michel
On Thu, Apr 3, 2014 at 6:05 AM, Sean Devoy <sdevoy@bizfocused.com> wrote: <image001.gif> Hi All,
I have a user that has claimed that if she picks up incoming calls, the audio is excellent. However, if someone else picks up the call and transfers it to her, the audio is "terrible".
While investigating I found that they are using their own custom MOH file. I suspect that the AUDIO configuration is being modified by the MOH feature and leaving her transferred call in a degraded state. But, I don't understand enough to actually see if that is the case.
Please review the pastebin file: http://pastebin.freeswitch.org/22302
I think I have trimmed it down to the relevant data. Ext 302 has answered and is just initiating the transfer.
If I am correct in my hypothesis, what is the correct solution?
Thanks in advance,
Sean
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
Try disabling G722, and only allow one single codec PCMU, see if the problem goes away. -- 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 3, 2014, at 12:25 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hey Brian,
All Cisco 504G phones.
Sean
I tried it Brian. Primary is PCMU, and Require Primary is "yes". Customer says still degraded. Other ideas? Thanks, Sean -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, April 03, 2014 1:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file Try disabling G722, and only allow one single codec PCMU, see if the problem goes away. -- 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 3, 2014, at 12:25 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hey Brian,
All Cisco 504G phones.
Sean
can you further characterize degraded. is it packet loss? is it a lesser quality codec? On Fri, Apr 4, 2014 at 12:46 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
I tried it Brian. Primary is PCMU, and Require Primary is "yes".
Customer says still degraded.
Other ideas?
Thanks, Sean
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, April 03, 2014 1:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file
Try disabling G722, and only allow one single codec PCMU, see if the problem goes away.
-- 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 3, 2014, at 12:25 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hey Brian,
All Cisco 504G phones.
Sean
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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What does degraded mean, try to describe it. -- 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 4, 2014, at 11:46 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
degraded.
Thanks Russell and Brian. I have requested details. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 04, 2014 1:54 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file What does degraded mean, try to describe it. -- 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 4, 2014, at 11:46 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
degraded.
Customers get mad when you ask the same exact question ten different ways! :P But doing so lets me figure out what exactly is going on most of the time. -- 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 4, 2014, at 2:02 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Thanks Russell and Brian.
I have requested details.
They are not a font of information! When I asked for more specific details regarding the degradation their COMPLETE response was: " The call sounds like I'm "under water", thats how most of them describe it!" I have asked if it is degraded in both directions. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 04, 2014 1:54 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file What does degraded mean, try to describe it. -- 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 4, 2014, at 11:46 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
degraded.
What endpoint was this? -- 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 7, 2014, at 10:34 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
They are not a font of information!
When I asked for more specific details regarding the degradation their COMPLETE response was: " The call sounds like I'm "under water", thats how most of them describe it!"
I have asked if it is degraded in both directions.
An external IP call comes in, is answered by a user on Cisco SPA504g. Transferred (with user made MOH audio playing) to another user on a Cisco SPA504G with Primary Codec PCMU and require primary = true. Does that help? Sean -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, April 07, 2014 5:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file What endpoint was this? -- 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 7, 2014, at 10:34 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
They are not a font of information!
When I asked for more specific details regarding the degradation their COMPLETE response was: " The call sounds like I'm "under water", thats how most of them describe it!"
I have asked if it is degraded in both directions.
could this be the problem? [WARNING] mod_sofia.c:1281 Asynchronous PTIME not supported, changing our end from 30 to 20 could this be the problem "Asynchronous PTIME not supported, changing our end from 30 to 20" (line 444 of the provided trace)? On Mon, Apr 7, 2014 at 6:05 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
An external IP call comes in, is answered by a user on Cisco SPA504g. Transferred (with user made MOH audio playing) to another user on a Cisco SPA504G with Primary Codec PCMU and require primary = true.
Does that help?
Sean
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, April 07, 2014 5:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file
What endpoint was this? -- 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 7, 2014, at 10:34 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
They are not a font of information!
When I asked for more specific details regarding the degradation their COMPLETE response was: " The call sounds like I'm "under water", thats how most of them describe it!"
I have asked if it is degraded in both directions.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Russell, Thanks for trying. I would love to know how to fix that, I have always had that WARNING, but don't know where to set PTIME on the phones or in FS. However, since I get that regardless incoming call or a transferred call, it seems unlikely. Again, thanks for looking. Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Russell Treleaven Sent: Monday, April 07, 2014 7:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file could this be the problem? [WARNING] mod_sofia.c:1281 Asynchronous PTIME not supported, changing our end from 30 to 20 could this be the problem "Asynchronous PTIME not supported, changing our end from 30 to 20" (line 444 of the provided trace)? On Mon, Apr 7, 2014 at 6:05 PM, Sean Devoy <sdevoy@bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote: An external IP call comes in, is answered by a user on Cisco SPA504g. Transferred (with user made MOH audio playing) to another user on a Cisco SPA504G with Primary Codec PCMU and require primary = true. Does that help? Sean -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Brian West Sent: Monday, April 07, 2014 5:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file What endpoint was this? -- Brian West brian@freeswitch.org<mailto: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 7, 2014, at 10:34 AM, Sean Devoy <sdevoy@bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote:
They are not a font of information!
When I asked for more specific details regarding the degradation their COMPLETE response was: " The call sounds like I'm "under water", thats how most of them describe it!"
I have asked if it is degraded in both directions.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You need to set the phone to 0.020 for the RTP time its documented on the wiki in the spa devices section. I thought we used to auto fix this clearly they’ve messed it up again. -- 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 7, 2014, at 7:00 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi Russell,
Thanks for trying. I would love to know how to fix that, I have always had that WARNING, but don’t know where to set PTIME on the phones or in FS.
However, since I get that regardless incoming call or a transferred call, it seems unlikely.
Again, thanks for looking. Sean
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Russell Treleaven Sent: Monday, April 07, 2014 7:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file
could this be the problem? [WARNING] mod_sofia.c:1281 Asynchronous PTIME not supported, changing our end from 30 to 20
could this be the problem "Asynchronous PTIME not supported, changing our end from 30 to 20" (line 444 of the provided trace)?
On Mon, Apr 7, 2014 at 6:05 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: An external IP call comes in, is answered by a user on Cisco SPA504g. Transferred (with user made MOH audio playing) to another user on a Cisco SPA504G with Primary Codec PCMU and require primary = true.
Does that help?
Sean
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, April 07, 2014 5:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file
What endpoint was this? -- 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 7, 2014, at 10:34 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
They are not a font of information!
When I asked for more specific details regarding the degradation their COMPLETE response was: " The call sounds like I'm "under water", thats how most of them describe it!"
I have asked if it is degraded in both directions.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Hi Brian, I did not find RPT Packet Size anywhere in the SPA3xx/5xx section of the Wiki. I will update the Wiki now that I did find it on the phone. However, the phone defaults to .020. All (or almost all) of our phones are Cisco SPAs. Can't I just set the FS end to .020 to match all my devices? If so where? With regard to the original problem, I have tested. 203 is the problem phone. I called 203 directly - perfect results. I called 200 and was transferred to 203 - horrible audio. Not static, not popping, not dropped packets, just like someone with the mic in their mouth. Also, it APPEARED that the audio was quite delayed based on us talking on top of each other, but I could not confirm for 100% certainty. I then called 203 and was placed on hold. Audio remain great. I called 203 and was transferred to 200, audio remained great. So, my current findings are transfers to 203 and ONLY transfers to ONLY 203 ALWAYS have bad audio. I will be reviewing every item in 203s phone and FS configuration later this afternoon. Thanks, Sean -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, April 07, 2014 8:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file You need to set the phone to 0.020 for the RTP time its documented on the wiki in the spa devices section. I thought we used to auto fix this clearly they've messed it up again. -- 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 7, 2014, at 7:00 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi Russell,
Thanks for trying. I would love to know how to fix that, I have always had that WARNING, but don't know where to set PTIME on the phones or in FS.
However, since I get that regardless incoming call or a transferred call, it seems unlikely.
Again, thanks for looking. Sean
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Russell Treleaven Sent: Monday, April 07, 2014 7:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file
could this be the problem? [WARNING] mod_sofia.c:1281 Asynchronous PTIME not supported, changing our end from 30 to 20
could this be the problem "Asynchronous PTIME not supported, changing our end from 30 to 20" (line 444 of the provided trace)?
On Mon, Apr 7, 2014 at 6:05 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: An external IP call comes in, is answered by a user on Cisco SPA504g. Transferred (with user made MOH audio playing) to another user on a Cisco SPA504G with Primary Codec PCMU and require primary = true.
Does that help?
Sean
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, April 07, 2014 5:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file
What endpoint was this? -- 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 7, 2014, at 10:34 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
They are not a font of information!
When I asked for more specific details regarding the degradation their COMPLETE response was: " The call sounds like I'm "under water", thats how most of them describe it!"
I have asked if it is degraded in both directions.
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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
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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
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. 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: Friday, April 11, 2014 1:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file 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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
This link should be informative http://www.opus-codec.org/comparison/ G722 uses the same amount of bandwidth as PCMU/PCMA. On Fri, Apr 11, 2014 at 1:35 PM, Moishe Grunstein <max@nysolutions.com>wrote:
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.
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: Friday, April 11, 2014 1:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file
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@lists.freeswitch.org [mailto: freeswitch-users-bounces@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@30ion 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
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks Russell and Moishe, 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." 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. Moishe has declined the bounty in favor of a donation the FS Dev's in his name. I will send it shortly. Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Russell Treleaven Sent: Friday, April 11, 2014 2:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file This link should be informative http://www.opus-codec.org/comparison/ G722 uses the same amount of bandwidth as PCMU/PCMA. On Fri, Apr 11, 2014 at 1:35 PM, Moishe Grunstein <max@nysolutions.com<mailto:max@nysolutions.com>> wrote: 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. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Sean Devoy Sent: Friday, April 11, 2014 1:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file 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@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@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@30i on your profile or for that user. -- Brian West brian@freeswitch.org<mailto: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<mailto: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<http://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<http://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<http://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<http://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<http://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> [mailto: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<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto: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> [mailto: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
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Was he using a gui and if so I wonder which one? I haven't seen any that will choose the codec like that. Sean Devoy <sdevoy@bizfocused.com> wrote:
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
Sorry, I should have been more clear. I was referring to the GUI WEB configuration pages on the CISCO SPA504g itself. Sean -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of covici@ccs.covici.com Sent: Saturday, April 12, 2014 1:47 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH file Was he using a gui and if so I wonder which one? I haven't seen any that will choose the codec like that. Sean Devoy <sdevoy@bizfocused.com> wrote:
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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
____________________________________________________________________ __ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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____________________________________________________________________ __ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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____________________________________________________________________ __ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
It appears that the user has been "helping". I have reset the admin password. The primary codec had been reset to PCMU, AGAIN. I have reset them all to G722 and disabled all others. I am waiting for results of their tests. 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (6)
-
Brian West
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covici＠ccs.covici.com
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Michel Brabants
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Moishe Grunstein
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Russell Treleaven
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Sean Devoy