Re: [asterisk-users] Meetme voice quality problems
Matthew J. Roth a écrit : Administrator TOOTAI wrote: This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards Daniel, I thought that using an empty TDM400P as a timing source may no longer be the best solution due to the emergence of new stable timing sources (such as HPET), but this is an interesting issue. Are you stating that you can't put an X100P or a TDM400P with no lines attached alongside a B410P because it impacts the stability of Asterisk? Yes Do you have any idea why? No Can't the B410P be used as a timing source? No What have you done to provide stable timing? ztdummy, not always stable :-( I know that's a lot of questions, but I'm genuinely curious. ;-) It seems very strange that a TDM400P in timingonly mode and no lines attached would have any impact on Asterisk's stability. I have to add that this is mainly true with 2 B410P in the server or with B410P in amd64 machines. Seems also that Debian Etch with a 2.6.18 kernel is not the best :-( -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Ubuntu has a real time kernel in repository apt-get install linux-rt . So you dont need to recompile . I think debian should also have one in repository , or u can manually compile a real time enabled kernel . Here's what is shows with real time patched kernel . dmesg|grep ztdummy [ 53.293071] ztdummy: Trying to load High Resolution Timer [ 53.293076] ztdummy: Initialized High Resolution Timer [ 53.293078] ztdummy: Starting High Resolution Timer [ 53.293080] ztdummy: High Resolution Timer started, good to go zttest Opened pseudo zap interface, measuring accuracy... 100.00% 99.987793% 99.792480% 99.780273% 99.987793% 99.975586% 99.987793% 99.987793% 100.00% 100.00% 100.00% 99.987793% 99.987793% 99.987793% 100.00% 100.00% 99.987793% 100.00% --- Results after 18 passes --- Best: 100.00 -- Worst: 99.780273 -- Average: 99.969482 On Feb 2, 2008 10:27 PM, Administrator TOOTAI [EMAIL PROTECTED] wrote: Matthew J. Roth a écrit : Administrator TOOTAI wrote: This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards Daniel, I thought that using an empty TDM400P as a timing source may no longer be the best solution due to the emergence of new stable timing sources (such as HPET), but this is an interesting issue. Are you stating that you can't put an X100P or a TDM400P with no lines attached alongside a B410P because it impacts the stability of Asterisk? Yes Do you have any idea why? No Can't the B410P be used as a timing source? No What have you done to provide stable timing? ztdummy, not always stable :-( I know that's a lot of questions, but I'm genuinely curious. ;-) It seems very strange that a TDM400P in timingonly mode and no lines attached would have any impact on Asterisk's stability. I have to add that this is mainly true with 2 B410P in the server or with B410P in amd64 machines. Seems also that Debian Etch with a 2.6.18 kernel is not the best :-( -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Matthew J. Roth a écrit : [...] I settled on using an empty TDM400P as a timing source, because it is a simple solution that just works. This may still be your best bet, but I'll defer judgment on that to the list because Asterisk has evolved quite a bit since I made that decision. This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Administrator TOOTAI wrote: This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards Daniel, I thought that using an empty TDM400P as a timing source may no longer be the best solution due to the emergence of new stable timing sources (such as HPET), but this is an interesting issue. Are you stating that you can't put an X100P or a TDM400P with no lines attached alongside a B410P because it impacts the stability of Asterisk? Do you have any idea why? Can't the B410P be used as a timing source? What have you done to provide stable timing? I know that's a lot of questions, but I'm genuinely curious. It seems very strange that a TDM400P in timingonly mode and no lines attached would have any impact on Asterisk's stability. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
On Jan 30, 2008 10:35 PM, Dan Austin [EMAIL PROTECTED] wrote: Franklin wrote: ztdummy can give you issues as a timing device. Yes and no. See below Any way you could try using a Digium card just as a timing device to see if this helps? Tomasz wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. Your kernel is new enough that you should be able to leverage hi-res timers (you might need to patch ztdummy), or at least a RTC set to 8192 ticks/sec. What does dmesg show after ztdummy is loaded? it is 1024 Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.4-r3748 Zaptel Echo Canceller: MG2 ztdummy: RTC rate is 1024 how can I increase it? I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Do you have internal_timing=yes in asterisk.conf? This option allows Asterisk to time the RTP stream based on zaptel/ztdummy clock and not on the received RTP stream. In a MeetMe, where callers might mute themselves, the received RTP stream is all but useless for timing. Yes I have it set. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
On Jan 30, 2008 5:48 PM, Matthew J. Roth [EMAIL PROTECTED] wrote: Tomasz Zieleniewski wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? Tomasz, Have you run zttest on the system? It verifies the accuracy of your timing source. Digium recommends an accuracy of at least 99.98%. If your accuracy is less than that it's probably the source of your problem. ztttest results show value below 99,98: [EMAIL PROTECTED]:~/src/zaptel-1.4$ ./zttest -v -c 5 Opened pseudo zap interface, measuring accuracy... 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4064.232 system clock sample intervals (49.612%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.240 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4064.232 system clock sample intervals (49.612%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.232 system clock sample intervals (50.003%) 8192 zaptel samples in 4096.240 system clock sample intervals (50.003%) --- Results after 11 passes --- Best: 50.003 -- Worst: 49.612 -- Average: 49.931827, Difference: 49.931827 Luckily, it's a problem with multiple solutions. The following thread documents some kernel configuration changes that you can make to improve the quality of ztdummy as a timing source: Recommendations for kernel config http://lists.digium.com/pipermail/asterisk-users/2007-October/197778.html Do You know how can I check and set kernel timer frequency? My preferred solution is to use an empty TDM400P as a timing source. It will cost you a little bit of money, but it's an easy way to reliably solve your problem. You'll find a few posts about it if you search the list, but this one has most of the information you'll need: Empty Wildcard TDM400P as a MeetMe timer. http://lists.digium.com/pipermail/asterisk-users/2007-March/182005.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Tomasz Zieleniewski wrote: ztttest results show value below 99,98: [EMAIL PROTECTED]:~/src/zaptel-1.4$ ./zttest -v -c 5 snip --- Results after 11 passes --- Best: 50.003 -- Worst: 49.612 -- Average: 49.931827, Difference: 49.931827 This is the first thing I would address. Get that average to at least 99.98% and it's likely that your problem will go away. Do You know how can I check and set kernel timer frequency? You can check the timer frequency as follows: # grep -e ^CONFIG_HZ /boot/config-`uname -r` CONFIG_HZ_1000=y CONFIG_HZ=1000 Setting it requires configuring and rebuilding the kernel. Try setting CONFIG_HZ=1000 and checking the results of zttest. Depending on how new your kernel is there are more options, but this is a good place to start. I settled on using an empty TDM400P as a timing source, because it is a simple solution that just works. This may still be your best bet, but I'll defer judgment on that to the list because Asterisk has evolved quite a bit since I made that decision. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme voice quality problems
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is cut.
Each voice sequence is disturbed.
Does any one have similar issue and could give me some advice??
my extension.conf for meetme:
;switch = Realtime/macro-conference
exten = s,1,NoOp(-- Macro-conference for:${MARCO_EXTEN} start --)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,MeetMe(|cdIps)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
Thank for any help.
Kind Regards
Tomasz
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Re: [asterisk-users] Meetme voice quality problems
Tomasz Zieleniewski wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? Tomasz, Have you run zttest on the system? It verifies the accuracy of your timing source. Digium recommends an accuracy of at least 99.98%. If your accuracy is less than that it's probably the source of your problem. Luckily, it's a problem with multiple solutions. The following thread documents some kernel configuration changes that you can make to improve the quality of ztdummy as a timing source: Recommendations for kernel config http://lists.digium.com/pipermail/asterisk-users/2007-October/197778.html My preferred solution is to use an empty TDM400P as a timing source. It will cost you a little bit of money, but it's an easy way to reliably solve your problem. You'll find a few posts about it if you search the list, but this one has most of the information you'll need: Empty Wildcard TDM400P as a MeetMe timer. http://lists.digium.com/pipermail/asterisk-users/2007-March/182005.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
ztdummy can give you issues as a timing device. Any way you could try using a
Digium card just as a timing device to see if this helps?
- Original Message -
From: Tomasz Zieleniewski [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 30, 2008 11:23:57 AM (GMT-0500) America/New_York
Subject: [asterisk-users] Meetme voice quality problems
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is cut.
Each voice sequence is disturbed.
Does any one have similar issue and could give me some advice??
my extension.conf for meetme:
;switch = Realtime/macro-conference
exten = s,1,NoOp(-- Macro-conference for:${MARCO_EXTEN} start --)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,MeetMe(|cdIps)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
Thank for any help.
Kind Regards
Tomasz
--
Franklin Webb
Asst Project Manager
Inter Medi@ Marketing Solutions
610-701-9670
[EMAIL PROTECTED]
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Re: [asterisk-users] Meetme voice quality problems
Franklin wrote: ztdummy can give you issues as a timing device. Yes and no. See below Any way you could try using a Digium card just as a timing device to see if this helps? Tomasz wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. Your kernel is new enough that you should be able to leverage hi-res timers (you might need to patch ztdummy), or at least a RTC set to 8192 ticks/sec. What does dmesg show after ztdummy is loaded? I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Do you have internal_timing=yes in asterisk.conf? This option allows Asterisk to time the RTP stream based on zaptel/ztdummy clock and not on the received RTP stream. In a MeetMe, where callers might mute themselves, the received RTP stream is all but useless for timing. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
