Hello! On a box with a Sangoma A500 BRI card i have a serious audio quality problem. The "outside"-leg of a call hears a choppy robotic noise, it's difficult to understand the actual voice. I hope someone recognizes this issue, as i have tried several approaches and so far i neither know what causes the problem nor where it is caused exactly. Here's a sample file(The noise is not very noticeable, it's the oscillating hum you hear in the background, the short voice-sample you hear is mine, but it's inbound so it's not affected): http://poab.org/test1.wav If the call is set on-hold or if there's a playback/moh, the quality for the outside-leg is fine while there's no two-way-audio, so my assumption was that it's some kind of problem with codecs or transcoding. Capturing the RTP-Stream from the phone to the server proved though that the quality is fine before it arrives at FS(There are no audio problems on internal calls as well). Here's what else i've tried without success: - set disable-monotonic-timing to true - started FS with the -rt Flag(As well as -waste and -heavy-timer) - jitter "time_test" 100 100 responds with avg 130(Thats fine right?) - rearranged the codec-settings of the phone(Sangoma SPIP 450) to give A-law the highest priority - high-precision-timer is available in the kernel according to dmesg and /proc/timer_list (kernel is 2.6.32-5-686 on debian 6.0) Sangoma's Support appears to be clueless as well, they didn't have effective suggestions so far(Opened a ticket 2 weeks). Since i don't have this problem with identical hardware without Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. Probably an interrupt issue? Any suggestions would be highly appreciated, i'm a bit at loss now and only know more drastic next steps(Replacing ISDN-Modules or the whole hardware). Best regards, Christian
You should choose your subject more carefully mentioning Sangoma or FreeTDM in the subject to attract the attention of Moy On Thu, Mar 1, 2012 at 8:45 AM, Christian Benke <benkokakao@gmail.com> wrote:
Hello!
On a box with a Sangoma A500 BRI card i have a serious audio quality problem. The "outside"-leg of a call hears a choppy robotic noise, it's difficult to understand the actual voice. I hope someone recognizes this issue, as i have tried several approaches and so far i neither know what causes the problem nor where it is caused exactly.
Here's a sample file(The noise is not very noticeable, it's the oscillating hum you hear in the background, the short voice-sample you hear is mine, but it's inbound so it's not affected):
If the call is set on-hold or if there's a playback/moh, the quality for the outside-leg is fine while there's no two-way-audio, so my assumption was that it's some kind of problem with codecs or transcoding. Capturing the RTP-Stream from the phone to the server proved though that the quality is fine before it arrives at FS(There are no audio problems on internal calls as well).
Here's what else i've tried without success: - set disable-monotonic-timing to true - started FS with the -rt Flag(As well as -waste and -heavy-timer) - jitter "time_test" 100 100 responds with avg 130(Thats fine right?) - rearranged the codec-settings of the phone(Sangoma SPIP 450) to give A-law the highest priority - high-precision-timer is available in the kernel according to dmesg and /proc/timer_list (kernel is 2.6.32-5-686 on debian 6.0)
Sangoma's Support appears to be clueless as well, they didn't have effective suggestions so far(Opened a ticket 2 weeks). Since i don't have this problem with identical hardware without Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. Probably an interrupt issue?
Any suggestions would be highly appreciated, i'm a bit at loss now and only know more drastic next steps(Replacing ISDN-Modules or the whole hardware).
Best regards, Christian
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On Thu, Mar 1, 2012 at 9:45 AM, Christian Benke <benkokakao@gmail.com>wrote:
Sangoma's Support appears to be clueless as well, they didn't have effective suggestions so far(Opened a ticket 2 weeks). Since i don't have this problem with identical hardware without Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. Probably an interrupt issue?
Another test you can try to pinpoint the issue and determine once for all if the problem is below FreeSWITCH is to use the "ftdm trace" command. This command will create an input an output file (in the directory you specify) that contains the raw audio as it is read and written from/to the wanpipe device. Type "ftdm trace" to see the command syntax. Run the command whenever you reproduce the problem (can be run in the middle of the call) then stop the trace with "ftdm notrace" when you're done capturing. Note that the tracing will not stop automatically when the call is hung up, the trace is a raw I/O trace and has no knowledge of calls starting or being hung up. The audio format will be alaw (because is a BRI line), you can convert that to a wav (with sox for example) and see if the audio issue is there. If it is, we can start thinking on lower-level troubleshooting. *Moises Silva **Manager, Software Engineering*** msilva@sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 **<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`| | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
Christian, I have a similar problem with a Sangoma B700 card (I think the ISDN part is similar, though) and have a Sangoma ticket open (#1310). It's intermittent and relies on the outside party reporting the quality being poor, so I am waiting for another occasion to arise when I can grab some more detailed traces. Your sample is quiet, which makes it hard to see what is going on. I have generated a test tones file which I use as a voicemail outgoing message on a test extension which allows me to clearly see what is happening to the voice. It uses sox and goes like this: #!/bin/bash sox -r 8000 -n out300_0_2.wav synth 2 sine 300 sox -r 8000 -n out500_0_2.wav synth 2 sine 500 sox -r 8000 -n out1000_0_2.wav synth 2 sine 1000 sox -r 8000 -n out2000_0_2.wav synth 2 sine 2000 sox -r 8000 -n out3000_0_2.wav synth 2 sine 3000 sox -r 8000 -n out300-3400_0_3.wav synth 3 sine 300-3400 sox -r 8000 -n out1-4000_0_5.wav synth 5 sine 1-4000 sox -r 8000 -n out1000_0_5.wav synth 5 sine 1000 sox -r 8000 -n out1000_-6_5.wav synth 5 sine 1000 gain -6 sox -r 8000 -n out1000_-12_5.wav synth 5 sine 1000 gain -12 sox -r 8000 -n outwn_0_5.wav synth 5 noise sox -r 8000 -n outwn_-6_5.wav synth 5 noise gain -6 sox -r 8000 -n outwn_-12_5.wav synth 5 noise gain -12 sox -r 8000 -n outsilence_2.wav synth 2 sine 300 vol 0 amplitude sox out300_0_2.wav outsilence_2.wav out500_0_2.wav outsilence_2.wav out1000_0_2.wav outsilence_2.wav out2000_0_2.wav outsilence_2.wav out3000_0_2.wav outsilence_2.wav out300-3400_0_3.wav outsilence_2.wav out1-4000_0_5.wav outsilence_2.wav out1000_0_5.wav outsilence_2.wav out1000_-6_5.wav outsilence_2.wav out1000_-12_5.wav outsilence_2.wav outwn_0_5.wav outsilence_2.wav outwn_-6_5.wav outsilence_2.wav outwn_-12_5.wav testtones.wav You may find something similar useful, or you can download this from http://www.earthspike.net/FreeSWITCH/testtones.wav (2.2MB on my domestic broadband). This is what I have recorded so far (using another FreeSWITCH server on the PSTN side): http://www.earthspike.net/FreeSWITCH/choppy_voice.wav (3.3MB on domestic broadband). You will notice that the outgoing IVR message is perfectly clear, and the voicemail outgoing message (test tones) gets messed with. Both are 8kHz WAV, so there is no transcoding or anything like that going on. Sometimes hold music goes choppy, sometimes not, sometimes the IVR message goes choppy as well. Sometimes the choppiness persists until I restart FreeSWITCH; other times it is for just one call. I have also written a script to capture 15s of the voice legs, using capture commands outlined by Sangoma support to include stuff around the echo canceller. You may also find this useful: #!/bin/bash echo "Starting diagnostics" DIR_PREFIX=/usr/local/freeswitch/recordings/diagnostics FILE_PREFIX=${DIR_PREFIX}/`date +%Y%m%d-%H%M%S` FS_CLI_X="/usr/local/freeswitch/bin/fs_cli -x" WAN_EC_C=/usr/sbin/wan_ec_client SPAN=`${FS_CLI_X} 'show channels' | sed -e '/FreeTDM/!d;s|.*FreeTDM/\([0-9]\):\([0-9]\)/.*|\1|;'` CHNL=`${FS_CLI_X} 'show channels' | sed -e '/FreeTDM/!d;s|.*FreeTDM/\([0-9]\):\([0-9]\)/.*|\2|;'` echo Diagnostic capture utility echo ========================== echo Detected call on ${SPAN}:${CHNL} echo Setting up trace on span wp${SPAN} ${FS_CLI_X} "ftdm trace ${FILE_PREFIX} ${SPAN}" pushd ${DIR_PREFIX} echo Recording 15s EC sample on ${SPAN}:${CHNL} ${WAN_EC_C} wanpipe${SPAN} monitor ${CHNL} popd echo Stopping trace ${FS_CLI_X} "ftdm notrace ${SPAN}" echo Done. The script relies on there only being one channel active when you want to record the call, so you may have to tune this for your requirements. I have created a folder recordings/diagnostics for the captured output. The non-profit I do this for has low enough traffic to make the single call assumption usually valid. If I am near an ssh session, though, I would work out the SPAN/CHNL variables manually in case more than one was active. If you get anywhere with this, let me or the list know. Note for devs: I haven't opened a FreeSWITCH JIRA on this because it is only calls involving my Sangoma card that are affected. If the deeper tracing that Sangoma have requested show that it is not in their card, then it will go to the FreeSWITCH JIRA. If you wish, I can open a JIRA but I'm not sure it's a good thing to have 2 separate fixes going on. John On 01/03/12 16:45, Moises Silva wrote:
On Thu, Mar 1, 2012 at 9:45 AM, Christian Benke <benkokakao@gmail.com <mailto:benkokakao@gmail.com>> wrote:
Sangoma's Support appears to be clueless as well, they didn't have effective suggestions so far(Opened a ticket 2 weeks). Since i don't have this problem with identical hardware without Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. Probably an interrupt issue?
Another test you can try to pinpoint the issue and determine once for all if the problem is below FreeSWITCH is to use the "ftdm trace" command. This command will create an input an output file (in the directory you specify) that contains the raw audio as it is read and written from/to the wanpipe device.
Type "ftdm trace" to see the command syntax. Run the command whenever you reproduce the problem (can be run in the middle of the call) then stop the trace with "ftdm notrace" when you're done capturing. Note that the tracing will not stop automatically when the call is hung up, the trace is a raw I/O trace and has no knowledge of calls starting or being hung up.
The audio format will be alaw (because is a BRI line), you can convert that to a wav (with sox for example) and see if the audio issue is there. If it is, we can start thinking on lower-level troubleshooting.
*Moises Silva **/Manager, Software Engineering/***
msilva@sangoma.com <mailto:msilva@sangoma.com>
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
t. +1 800 388 2475 (N. America)
t. +1 905 474 1990 x128
f. +1 905 474 9223
** <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
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Hi John! Thanks for your long mail, it's always good to hear that you're not alone with a problem :-)
It's intermittent and relies on the outside party reporting the quality being poor
Exactly that! In my case though it's callers who annoyedly complain about the poor quality ;-)
Your sample is quiet, which makes it hard to see what is going on.
Yeah well, i recorded it with a call to a phone set to auto-answer as the problem is already audible enough on a "quiet" call. I couldn't reproduce it so far with hold music or a playback. What do you mean by "voicemail outgoing message"? Thanks for the soundfile and script-examples too, will have to intensify my monitoring as well... Best regards, Christian
On 01/03/12 20:38, Christian Benke wrote:
Your sample is quiet, which makes it hard to see what is going on. Yeah well, i recorded it with a call to a phone set to auto-answer as the problem is already audible enough on a "quiet" call. I couldn't reproduce it so far with hold music or a playback. What do you mean by "voicemail outgoing message"? I set up a test extension that has no registered handset/soft client, so calls go straight to voicemail. The sound file replaces greeting_1.wav in storage/voicemail/.../1999/ (or whatever number your test extension is).
John
I set up a test extension that has no registered handset/soft client, so calls go straight to voicemail. The sound file replaces greeting_1.wav in storage/voicemail/.../1999/ (or whatever number your test extension is).
Ah, ok - but wouldn't playback do the same?
On 1 March 2012 17:45, Moises Silva <moises.silva@gmail.com> wrote:
Another test you can try to pinpoint the issue and determine once for all if the problem is below FreeSWITCH is to use the "ftdm trace" command.
Thanks for the suggestion. I have taken a more drastic approach today and temporarily replaced the A500 with a OpenVOX-card. Will have to wait for results till there is more traffic tomorrow. For some reason it seems the broken audio does appear more often from morning to noon when there's more traffic, but does not depend on the number of concurrent calls(It also appears around noon when there are no other calls, but not in the evening when there's been no traffic for a while). However, this observation is not empiric. Cheers, Christian
Hi Chirstian, Have you checked your physical installation? The problem could be originated by an electrical peek or perhaps due to your equipment are not well isolated electrically (ground, peeks, etc). Perhaps an equipment or devise in your office introduce some noise when it is activated. You need do more monitoring. After change tdm board have you notice difference, the issue persist? On 3/1/2012 3:36 PM, Christian Benke wrote:
On 1 March 2012 17:45, Moises Silva<moises.silva@gmail.com> wrote:
Another test you can try to pinpoint the issue and determine once for all if the problem is below FreeSWITCH is to use the "ftdm trace" command. Thanks for the suggestion. I have taken a more drastic approach today and temporarily replaced the A500 with a OpenVOX-card. Will have to wait for results till there is more traffic tomorrow. For some reason it seems the broken audio does appear more often from morning to noon when there's more traffic, but does not depend on the number of concurrent calls(It also appears around noon when there are no other calls, but not in the evening when there's been no traffic for a while). However, this observation is not empiric.
Cheers, Christian
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On 2 March 2012 14:13, Saugort Dario Garcia Tovar <dgarcia@anew.com.ve> wrote:
Have you checked your physical installation? The problem could be originated by an electrical peek or perhaps due to your equipment are not well isolated electrically (ground, peeks, etc). Perhaps an equipment or devise in your office introduce some noise when it is activated.
I also have this suspicion, as it appears to happen more often at a certain time of the day.
You need do more monitoring.
After change tdm board have you notice difference, the issue persist?
It was better, but today there was not much traffic - will have to wait till Monday, when business starts again... Regards, Christian
Hi Moises Silva, am new to freeswitch and am facing same noise issue when making call from ISDN (freetdm) to SIP UA. As per above mail, i have did ftdm trace and listen the clips. I found same noise in generated voice clip of line 2 (sip). am attached both audio files for your reference (ISDN line 1 and SIP line 2). Kindly let me know, what needs to do to rectify it. http://freeswitch-users.2379917.n2.nabble.com/file/n7546814/isdn_line1-in-s2... isdn_line1-in-s2c1.wav http://freeswitch-users.2379917.n2.nabble.com/file/n7546814/sip_line2-out-s2... sip_line2-out-s2c1.wav -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choopy-one-way-noise-tp7332765... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hi bawajan, I have listened to the files. What I can hear is a constant stationary noise. Usually the origin of such noise is hardware issues (mic/cables/etc.). If you are unable to find & fix the source of the problem, you can use noise cancellation software that will easily remove this noise. One software that you can try is PBXMate. A usage example is available on FreeSWITCH wiki: http://wiki.freeswitch.org/wiki/PBXMate-FreeSWITCH-integration Valer ________________________________ From: bawajan <shaik.bawajan@gmail.com> To: freeswitch-users@lists.freeswitch.org Sent: Thursday, May 10, 2012 5:02 PM Subject: Re: [Freeswitch-users] Choopy one-way noise Hi Moises Silva, am new to freeswitch and am facing same noise issue when making call from ISDN (freetdm) to SIP UA. As per above mail, i have did ftdm trace and listen the clips. I found same noise in generated voice clip of line 2 (sip). am attached both audio files for your reference (ISDN line 1 and SIP line 2). Kindly let me know, what needs to do to rectify it. http://freeswitch-users.2379917.n2.nabble.com/file/n7546814/isdn_line1-in-s2... isdn_line1-in-s2c1.wav http://freeswitch-users.2379917.n2.nabble.com/file/n7546814/sip_line2-out-s2... sip_line2-out-s2c1.wav -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choopy-one-way-noise-tp7332765... Sent from the freeswitch-users mailing list archive at Nabble.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
Hi Valer, Thanks, here am using a SIP UA (snom SI 120 IP Phone) and i have tested making calls from SIP UA to ISDN (freetdm). In this case the voice is very clear and good quality. Here am attaching both files for your reference. Kindly check it and help me to resolve. http://freeswitch-users.2379917.n2.nabble.com/file/n7549294/sip_line1-out-s2... sip_line1-out-s2c1.wav http://freeswitch-users.2379917.n2.nabble.com/file/n7549294/isdn_line2-in-s2... isdn_line2-in-s2c1.wav -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choopy-one-way-noise-tp7332765... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hi, Am using Sangoma card AFT-A108-SH with PRI lines -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choopy-one-way-noise-tp7332765... Sent from the freeswitch-users mailing list archive at Nabble.com.
participants (7)
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Anthony Minessale
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bawajan
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Christian Benke
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John
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Moises Silva
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Saugort Dario Garcia Tovar
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Valer Nur