Audio rate changes and clipping on playback
Hello all! I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host. My test setup is: Client <-> NAT ADSL <-> Freeswitch I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk. I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok. So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format: $ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others). Is anyone else seeing the same issues or has a solution? thanks, - Dale
Dale, What SVN rev are you running? Is this a xen or other virtual instance? I have done extensive testing in this area and haven't seen this... It could be related to QoS bits. Have you tried to use iptables to tag your traffic? Check this link http://wiki.freeswitch.org/wiki/QoS/ FreeSWITCH doesn't tag any traffic with any QoS bits and Asterisk does... So you might have to use the above URL to cause the traffic to get tagged and prioritized on the ADSL router. /b On Jan 26, 2008, at 7:29 AM, freeswitch@dalethatcher.com wrote:
Hello all!
I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host.
My test setup is:
Client <-> NAT ADSL <-> Freeswitch
I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk.
I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok.
So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format:
$ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others).
Is anyone else seeing the same issues or has a solution?
thanks,
- Dale
_______________________________________________ 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
Why is it that FS doesn't offer QoS packet tagging as a feature? It seems kind of kludgy to run another program on the same box to tag the packets before they are sent out. I know everything in FS is pretty well thought out, so I am sure there is a reason, I am just curious what it is. Peder Brian West wrote:
Dale, What SVN rev are you running? Is this a xen or other virtual instance? I have done extensive testing in this area and haven't seen this... It could be related to QoS bits. Have you tried to use iptables to tag your traffic? Check this link http://wiki.freeswitch.org/wiki/QoS/
FreeSWITCH doesn't tag any traffic with any QoS bits and Asterisk does... So you might have to use the above URL to cause the traffic to get tagged and prioritized on the ADSL router.
/b
On Jan 26, 2008, at 7:29 AM, freeswitch@dalethatcher.com wrote:
Hello all!
I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host.
My test setup is:
Client <-> NAT ADSL <-> Freeswitch
I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk.
I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok.
So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format:
$ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others).
Is anyone else seeing the same issues or has a solution?
thanks,
- Dale
_______________________________________________ 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
_______________________________________________ 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
Doing packet tagging in a cross platform way is a bit difficult, and we have just never tried to do it when there are other simple ways to handle. It would probably involve adding the ability to do qos tagging to APR. If someone is willing to get patches into apr for this, it would make sense to look at adding that support to freeswitch at some point in the future. Mike On Jan 26, 2008, at 4:10 PM, Peder @ NetworkOblivion wrote:
Why is it that FS doesn't offer QoS packet tagging as a feature? It seems kind of kludgy to run another program on the same box to tag the packets before they are sent out. I know everything in FS is pretty well thought out, so I am sure there is a reason, I am just curious what it is.
Peder
Brian West wrote:
Dale, What SVN rev are you running? Is this a xen or other virtual instance? I have done extensive testing in this area and haven't seen this... It could be related to QoS bits. Have you tried to use iptables to tag your traffic? Check this link http://wiki.freeswitch.org/wiki/QoS/
FreeSWITCH doesn't tag any traffic with any QoS bits and Asterisk does... So you might have to use the above URL to cause the traffic to get tagged and prioritized on the ADSL router.
/b
On Jan 26, 2008, at 7:29 AM, freeswitch@dalethatcher.com wrote:
Hello all!
I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host.
My test setup is:
Client <-> NAT ADSL <-> Freeswitch
I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk.
I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok.
So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format:
$ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others).
Is anyone else seeing the same issues or has a solution?
thanks,
- Dale
_______________________________________________ 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
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_______________________________________________ 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
Peder, APR doesn't give you access to the socket to do those operations. And I personally do not fell it belongs inside FreeSWITCH. Doing so with your routers and network infrastructure is the correct way to do it. We can do it with IPtables till such time someone writes a patch for APR to allow that to take place inside FreeSWITCH... And doing so in an abstract way that can work on every platform. /b On Jan 26, 2008, at 1:10 PM, Peder @ NetworkOblivion wrote:
Why is it that FS doesn't offer QoS packet tagging as a feature? It seems kind of kludgy to run another program on the same box to tag the packets before they are sent out. I know everything in FS is pretty well thought out, so I am sure there is a reason, I am just curious what it is.
Peder
Brian, Thanks for getting back to me. The SVN revision is 7343. The host is dedicated hardware (to be honest I didn't even think it would be ok to run freeswitch on a virtual host). I'll give the QoS settings a go and see if it clears up the problem. Could you send me a link to an audio file that is known to play fine? I will send you the SIP details of my server off list if you have a sec to see if you can replicate the issue. thanks, - Dale On Sat, Jan 26, 2008 at 11:33:01AM -0800, Brian West wrote:
Dale, What SVN rev are you running? Is this a xen or other virtual instance? I have done extensive testing in this area and haven't seen this... It could be related to QoS bits. Have you tried to use iptables to tag your traffic? Check this link http://wiki.freeswitch.org/wiki/QoS/
FreeSWITCH doesn't tag any traffic with any QoS bits and Asterisk does... So you might have to use the above URL to cause the traffic to get tagged and prioritized on the ADSL router.
/b
On Jan 26, 2008, at 7:29 AM, freeswitch@dalethatcher.com wrote:
Hello all!
I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host.
My test setup is:
Client <-> NAT ADSL <-> Freeswitch
I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk.
I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok.
So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format:
$ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others).
Is anyone else seeing the same issues or has a solution?
thanks,
- Dale
_______________________________________________ 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
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On Jan 26, 2008, at 11:27 PM, freeswitch@dalethatcher.com wrote:
Thanks for getting back to me.
The SVN revision is 7343. The host is dedicated hardware (to be honest I didn't even think it would be ok to run freeswitch on a virtual host).
Yep it works fine in Xen.
I'll give the QoS settings a go and see if it clears up the problem. Could you send me a link to an audio file that is known to play fine?
make sounds-music-8000-install then dial 9999
I will send you the SIP details of my server off list if you have a sec to see if you can replicate the issue.
I don't think i'll be able to replicate this one. But I can try when I get home tomorrow. /b
don't know what you mean with clipping, but I've notice clipping errors when I play sound files with maximum levels with G.711, clipping means that on max. posivite value a max. negative value is played. this is true only on G.711, e. g. Linear16 codes are fine. regards, thomas -----Ursprüngliche Nachricht----- Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von freeswitch@dalethatcher.com Gesendet: Samstag, 26. Januar 2008 16:30 An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] Audio rate changes and clipping on playback Hello all! I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host. My test setup is: Client <-> NAT ADSL <-> Freeswitch I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk. I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok. So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format: $ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others). Is anyone else seeing the same issues or has a solution? thanks, - Dale _______________________________________________ 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
I mean it sounds like parts of the sound are missing. e.g. the frequency of playback seems to change and I can hear a warbling/slow down from the Freeswitch instance on the audio samples. If you (or any others on the list) have a spare couple of minutes then I'm happy to send you the details of my test server. It would be reassuring if someone else could hear and verify the issue. thanks, - Dale On Mon, Jan 28, 2008 at 02:05:20PM +0100, Sluschny, Thomas wrote:
don't know what you mean with clipping, but I've notice clipping errors when I play sound files with maximum levels with G.711, clipping means that on max. posivite value a max. negative value is played. this is true only on G.711, e. g. Linear16 codes are fine.
regards, thomas
-----Ursprüngliche Nachricht----- Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von freeswitch@dalethatcher.com Gesendet: Samstag, 26. Januar 2008 16:30 An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] Audio rate changes and clipping on playback
Hello all!
I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host.
My test setup is:
Client <-> NAT ADSL <-> Freeswitch
I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk.
I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok.
So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format:
$ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others).
Is anyone else seeing the same issues or has a solution?
thanks,
- Dale
_______________________________________________ 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
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I have called your public extensions you have and I don't hear this issue. Let me do some more digging. /b On Jan 28, 2008, at 8:01 AM, freeswitch@dalethatcher.com wrote:
I mean it sounds like parts of the sound are missing. e.g. the frequency of playback seems to change and I can hear a warbling/slow down from the Freeswitch instance on the audio samples.
If you (or any others on the list) have a spare couple of minutes then I'm happy to send you the details of my test server. It would be reassuring if someone else could hear and verify the issue.
thanks,
- Dale
Problem solved! Thanks to the efforts of Brian and the usual suspects on the mailing list the issue has been tracked down to comfort noise generation. Mike suggested adding the following to my conf/sip_profiles/default.xml in the 'settings' section: <param name="supress-cng" value="true"/> This cleared up the audio problems with both Gizmo and FWD Communicator. I'll add some notes to the wiki. thanks again, - Dale On Mon, Jan 28, 2008 at 09:03:43AM -0600, Brian West wrote:
I have called your public extensions you have and I don't hear this issue. Let me do some more digging.
/b
On Jan 28, 2008, at 8:01 AM, freeswitch@dalethatcher.com wrote:
I mean it sounds like parts of the sound are missing. e.g. the frequency of playback seems to change and I can hear a warbling/slow down from the Freeswitch instance on the audio samples.
If you (or any others on the list) have a spare couple of minutes then I'm happy to send you the details of my test server. It would be reassuring if someone else could hear and verify the issue.
thanks,
- Dale
_______________________________________________ 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 is probably also a 'good thing' for anyone dealing with asterisk... Asterisk does not have CNG support and you can notice some serious issues with the media when that happens... This is due to the way asterisk handles the RTP stream and is required to rcv media before it will transmit media after the call is established K
From: <freeswitch@dalethatcher.com> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Mon, 28 Jan 2008 16:10:15 +0000 To: <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Audio rate changes and clipping on playback
Problem solved!
Thanks to the efforts of Brian and the usual suspects on the mailing list the issue has been tracked down to comfort noise generation.
Mike suggested adding the following to my conf/sip_profiles/default.xml in the 'settings' section:
<param name="supress-cng" value="true"/>
This cleared up the audio problems with both Gizmo and FWD Communicator.
I'll add some notes to the wiki.
thanks again,
- Dale
On Mon, Jan 28, 2008 at 09:03:43AM -0600, Brian West wrote:
I have called your public extensions you have and I don't hear this issue. Let me do some more digging.
/b
On Jan 28, 2008, at 8:01 AM, freeswitch@dalethatcher.com wrote:
I mean it sounds like parts of the sound are missing. e.g. the frequency of playback seems to change and I can hear a warbling/slow down from the Freeswitch instance on the audio samples.
If you (or any others on the list) have a spare couple of minutes then I'm happy to send you the details of my test server. It would be reassuring if someone else could hear and verify the issue.
thanks,
- Dale
_______________________________________________ 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
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participants (6)
-
Brian West
-
freeswitch＠dalethatcher.com
-
Ken Rice
-
Michael Jerris
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Peder ＠ NetworkOblivion
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Sluschny, Thomas