Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On Sun, 2004-12-19 at 02:11, David Uzzell wrote: Then the other thing if mem serves me you are running 2.6 kernel so why not run ztdummy? With the 2.6 kernel this does not require any specialist Hardware or anything! Sorry, but maybe you should have read his posts more thoroughly. ztdummy is not an option because of his chipset. He has usb-ohci. ztdummy requires usb-uhci. Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Bruno Hertz wrote: On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote: I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. The problems you had and were resolved with ztdummy, were they primarily IAX related ? Since, after all, the main channels relying on special timers are Meetme, IAX and (maybe) MusicOnHold according to http://www.voip-info.org/wiki-Asterisk+timer Just want to be sure, since I still believe my mere demo playback issue likely has a different reason ... I'd like to chime in here as I have a similar problem. I have been toying with * on other (cheapo) hardware not so successfully (mainly due to the audio chipsets). I just purchased an ASUS AV8 (Socket 939 Athlon 64 3500+) system for my real world testing, it's a high end MB and overall it has 98% of the feature set for what I wanted to accomplish. Currently I'm running FreeBSD 5.3 under the amd64 port of the OS (fyi). I'm experiencing the exact same symptoms - choppy clicking of the demo voice. I'll start by saying that I have done a reasonable amount of research on *, MB chipsets, and FreeBSD, and I've spent considerable time getting the basic functionality to work. The ports version of * under FreeBSD needed some tweaking to work under amd64 vs i386, but I have a working version including h323 and oss that works with the demo stuff. From what I have read the issue with choppy sound under the demo voice seems to be due to a timing issue, one that can't be solved under FreeBSD with the zaprtp (linux) stuff, and I haven't seen anything as far as USB stuff that will handle this. I do not have a Digium card installed yet, but I will have a TDM400P in a couple of days. Will a Digium card with the current driver solve the problem ? (zaptel doesn't compile for FreeBSD 5.3 amd64, maybe for i386). Given that I have a working installation with the same symptoms as reported, I'm leaning towards us having the same problem. If this is a timing issue, it would be great to solve this in a systematic way (without external hardware). Thoughts? Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Martin List-Petersen wrote: On Sun, 2004-12-19 at 02:11, David Uzzell wrote: Then the other thing if mem serves me you are running 2.6 kernel so why not run ztdummy? With the 2.6 kernel this does not require any specialist Hardware or anything! Sorry, but maybe you should have read his posts more thoroughly. ztdummy is not an option because of his chipset. He has usb-ohci. ztdummy requires usb-uhci. Umm yes it does on 2.4 kernel but on a 2.6 kernel it doesn't cause I am running it on a 2.6 kernel and I don't have that hardware. Quoted from http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer On kernel version 2.6 it uses internal high-resolution kernel timer and do not require any additional hardware. Now in the original post he says that he is using FC2 so I am not 100% sure if it is 2.6 or 2.4 but FC2 is only one step away from FC3 which does run a 2.6 kernel. I don't know on FC2 as I have never run it. And yes to answer the original poster it did solve my IAX problems. With the demo I would sugest that maybe the SMP kernel on a single CPU server could be a partial cause. I have seen strange things on Dual CPU servers running SMP kernels were 1 CPU has been removed. Hope that helps. David Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. The problems you had and were resolved with ztdummy, were they primarily IAX related ? Since, after all, the main channels relying on special timers are Meetme, IAX and (maybe) MusicOnHold according to http://www.voip-info.org/wiki-Asterisk+timer Just want to be sure, since I still believe my mere demo playback issue likely has a different reason ... I'm 95% sure iax is not dependent on the ztdummy type timers. Maybe the OP could give us a little more detail on the specific data flow that he's having an issue with. I interpreted his call problem as: sipdev1 - ? - teliax.com - iax - OP-asterisk - sipdev2 He indicated sipdev1 was running VAD, and the call was completed via teliax.com to his asterisk with crackly audio. If this is the case, the issue is VAD between sipdev1 and the ? box shown in the data flow. Since there isn't a consistent flow of rtp data packets between sipdev1 and ? because of VAD, what gets sent to teliax.com is already choppy audio. There is nothing the OP is going to be able to fix between teliax.com and sipdev2 to correct for a problem that is located elsewhere. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
http://www.voip-info.org/wiki-RTP+Silence+Suppression http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html So I am confused. The first says that VAD is supported in RTP. Ok, I know that. The second is kinda ambiguous and seems to imply that * doesnt support VAD. I think it does now as there is a VAD=yes option in SIP.conf. Either way since IAX doesnt use RTP both statements are probably not relevant. Does * support VAD with IAX? If so can it be turned on and off in IAX?? Does anyone know definitively?? I really like to turn it off and just send packet continuously. Should I file a bug (feature request)?? Looking at the current /usr/src/asterisk/configs/sip.conf.sample, VAD=yes does not exist. Since those sample files tend to be the formal documentation for valid asterisk parameters, it should be safe to say its not supported. Same for iax.conf.sample; doesn't exist there either. The comment made by John Todd in the August 2003 posting was simply suggesting to the original poster (as that time) that he should enter a feature request into the asterisk bug tracker if he felt strongly that VAD was needed. The description of VAD in the voip-info reference is simply someone documenting what the sip rfc states about VAD. It does not imply or even suggest that asterisk supports VAD. Asterisk does not support VAD today (nor does it support every option documented in the sip rfc). The iax data flow betwen two boxes is not the same as sip-rtp data flows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On Sun, 2004-12-19 at 00:40 -0800, Chris Miller wrote: From what I have read the issue with choppy sound under the demo voice seems to be due to a timing issue Taking the risk of appearing notorious, I again emphasize that I don't believe that. I have asterisk right now with ztdummy running on a Debian Sarge box. When I connect with either GnomeMeeting/oh323 or iaxcomm via home LAN to that box, I experience exactly that symptoms. i.e. choppy demo voice. Now, if I boot FC3 on that same box, with the same asterisk version compiled under FC3, I *do not* get choppy sound even without ztdummy. Actually, I never bothered compiling zaptel support on FC3. Of course, I hope the card you're expecting will solve that problem for you, but I wouldn't be suprised if it didn't. Regards, Bruno. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On 19/12/2004 16:40 Chris Miller said the following: seems to be due to a timing issue, one that can't be solved under FreeBSD with the zaprtp (linux) stuff, and I haven't seen anything as the ztdummy pseudo timer works well under freebsd 4.x and 5.x. i used it for a bit before i got my digium cards. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On 19/12/2004 20:38 Rich Adamson said the following: I'm 95% sure iax is not dependent on the ztdummy type timers. trunked iax requires a timer, either ztdummy or a digium card. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Keith O'Brien wrote: I highly suggest you work on getting either the RTC or USB driver loaded to provide timing if you don't already have a PSTN card for that job. I am having a similar problem. Can someone point me to the procedure to install these virtual drivers for timing. I searched the wiki but came up empty. The URL you are looking for is: http://www.voip-info.org/wiki-Asterisk+timer Excerpt from that page: How to get a working timer * A Digium ZAPTEL INTERFACE has working timers. * If you don't have Digium hardware, there are two replacements: o ztdummy (in the standard zaptel distribution on the Asterisk CVS) on 2.4 Linux kernel uses USB-UHCI timers in USB drivers on platforms with UHCI USB support... o zaprtc found on http://www.junghanns.net/asterisk/ uses the real time clock in the PC instead... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On Sun, 2004-12-19 at 12:25 +1300, Matt Riddell wrote: o ztdummy (in the standard zaptel distribution on the Asterisk CVS) on 2.4 Linux kernel uses USB-UHCI timers in USB drivers on platforms with UHCI USB support... Uses USB on kernel 2.4, but not on 2.6. On the latter it's supposed to be a better rtc module. For 2.6 this driver is the only available option as far as I can see. o zaprtc found on http://www.junghanns.net/asterisk/ uses the real time clock in the PC instead... For 2.4 only. Regards, Bruno. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Citat Keith O'Brien [EMAIL PROTECTED]: The URL you are looking for is: http://www.voip-info.org/wiki-Asterisk+timer Thanks. After reading through the notes I checked my server (Dell 1750) and noted that it uses a USB OHCI interface so the first option doesn't appear to be an option. Also it indicates that the second option of using zaprtc http://www.voip-info.org/wiki-Asterisk+zaprtc won't work on SMP systems. The 1750 is a SMP system and I am running a 2.4 SMP kernel but do not actually have a second processor installed. Can I still use zaprtc with a SMP kernel if the second processor isn't actually installed?? zaprtc should work indeed if you only have one CPU in the system. Slán Leat, Martin List-Petersen Dublin, Eire (contact info == http://www.marlow.dk) -- We Klingons believe as you do -- the sick should die. Only the strong should live. -- Kras, Friday's Child, stardate 3497.2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Thanks. On a related note. The problem that I am troubleshooting has to do with an IAX connection to TELIAX. Outgoing calls are perfect. When I have incoming calls they are very crackly and break up. I have checked the jitter buffer and it is not overrunning so it doesn't appear to be a jitter or packet loss problem. I am beginning to suspect that since I don't have a ZAPtel card in my machine, * is losing sync with the incoming stream. From what I understand, if there isn't a zap timing source * uses the incoming data stream to derive timing. Since the incoming stream is using VAD, my assumption is that it is losing the timing during the pauses in the speech. Does anyone know of a way to just turn off VAD in *? This would have multiple benefits (if you have the bandwidth). Turning off VAD will improve voice quality by eliminating and front end clipping during talk spurts and I am assuming will also minimize the impact of not having a ZAP timing source. Is there a way to disable VAD in *? Thanks again. -Original Message- From: Martin List-Petersen [mailto:[EMAIL PROTECTED] Sent: Saturday, December 18, 2004 8:20 PM To: Keith O'Brien Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality Citat Keith O'Brien [EMAIL PROTECTED]: The URL you are looking for is: http://www.voip-info.org/wiki-Asterisk+timer Thanks. After reading through the notes I checked my server (Dell 1750) and noted that it uses a USB OHCI interface so the first option doesn't appear to be an option. Also it indicates that the second option of using zaprtc http://www.voip-info.org/wiki-Asterisk+zaprtc won't work on SMP systems. The 1750 is a SMP system and I am running a 2.4 SMP kernel but do not actually have a second processor installed. Can I still use zaprtc with a SMP kernel if the second processor isn't actually installed?? zaprtc should work indeed if you only have one CPU in the system. Slán Leat, Martin List-Petersen Dublin, Eire (contact info == http://www.marlow.dk) -- We Klingons believe as you do -- the sick should die. Only the strong should live. -- Kras, Friday's Child, stardate 3497.2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Keith O'Brien wrote: Thanks. On a related note. The problem that I am troubleshooting has to do with an IAX connection to TELIAX. Outgoing calls are perfect. When I have incoming calls they are very crackly and break up. I have checked the jitter buffer and it is not overrunning so it doesn't appear to be a jitter or packet loss problem. I am beginning to suspect that since I don't have a ZAPtel card in my machine, * is losing sync with the incoming stream. From what I understand, if there isn't a zap timing source * uses the incoming data stream to derive timing. I have been watching the to and fro of this over the last day or so and being a fairly newbie myself, Just looking plainly at what you have running, You have a SMP kernel running on a Dual Capable server but with only 1 cpu. Why Don't you run a NONE SMP kernel, one which would be suited to the fact that you only have one CPU in the server. Then the other thing if mem serves me you are running 2.6 kernel so why not run ztdummy? With the 2.6 kernel this does not require any specialist Hardware or anything! I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. David Since the incoming stream is using VAD, my assumption is that it is losing the timing during the pauses in the speech. Does anyone know of a way to just turn off VAD in *? This would have multiple benefits (if you have the bandwidth). Turning off VAD will improve voice quality by eliminating and front end clipping during talk spurts and I am assuming will also minimize the impact of not having a ZAP timing source. Is there a way to disable VAD in *? Thanks again. -Original Message- From: Martin List-Petersen [mailto:[EMAIL PROTECTED] Sent: Saturday, December 18, 2004 8:20 PM To: Keith O'Brien Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality Citat Keith O'Brien [EMAIL PROTECTED]: The URL you are looking for is: http://www.voip-info.org/wiki-Asterisk+timer Thanks. After reading through the notes I checked my server (Dell 1750) and noted that it uses a USB OHCI interface so the first option doesn't appear to be an option. Also it indicates that the second option of using zaprtc http://www.voip-info.org/wiki-Asterisk+zaprtc won't work on SMP systems. The 1750 is a SMP system and I am running a 2.4 SMP kernel but do not actually have a second processor installed. Can I still use zaprtc with a SMP kernel if the second processor isn't actually installed?? zaprtc should work indeed if you only have one CPU in the system. Slán Leat, Martin List-Petersen Dublin, Eire (contact info == http://www.marlow.dk) -- We Klingons believe as you do -- the sick should die. Only the strong should live. -- Kras, Friday's Child, stardate 3497.2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On Sunday 19 December 2004 02:00, Keith O'Brien wrote: Since the incoming stream is using VAD, my assumption is that it is losing the timing during the pauses in the speech. Does anyone know of a way to just turn off VAD in *? This would have multiple benefits (if you have the bandwidth). Turning off VAD will improve voice quality by eliminating and front end clipping during talk spurts and I am assuming will also minimize the impact of not having a ZAP timing source. Is there a way to disable VAD in *? It seems not: http://www.voip-info.org/wiki-RTP+Silence+Suppression http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html (I'm not sure if that second one has been superseded by more recent events? - but the first certainyl suggests that it's the sender which decides whether to use VAD or not, not the receiver) Antony. -- There's no such thing as bad weather - only the wrong clothes. - Billy Connolly Please reply to the list; please don't CC me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Thanks. On a related note. The problem that I am troubleshooting has to do with an IAX connection to TELIAX. Outgoing calls are perfect. When I have incoming calls they are very crackly and break up. I have checked the jitter buffer and it is not overrunning so it doesn't appear to be a jitter or packet loss problem. I am beginning to suspect that since I don't have a ZAPtel card in my machine, * is losing sync with the incoming stream. From what I understand, if there isn't a zap timing source * uses the incoming data stream to derive timing. The timing source that you're referring to is only needed for meetme conference and a couple of other items like that. Has nothing to do with iax. Since the incoming stream is using VAD, my assumption is that it is losing the timing during the pauses in the speech. Does anyone know of a way to just turn off VAD in *? This would have multiple benefits (if you have the bandwidth). Turning off VAD will improve voice quality by eliminating and front end clipping during talk spurts and I am assuming will also minimize the impact of not having a ZAP timing source. Is there a way to disable VAD in *? VAD would be the problem, and its not fixable at your * location since you're receiving iax packets from teliax.com (at least that was my understanding). The only solution you have is to convience the distant end to disable silence suppression. There has been some discussion on and off for some time about changing asterisk's VAD behaviour, but my understanding is that is a major effort and not sure where it stands. By the way, I'm a very happy camper with TELIAX.com and asterisk. Very clear and solid audio using iax-gsm in both directions. Excellent customer support to date, and I don't work for them. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote: I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. The problems you had and were resolved with ztdummy, were they primarily IAX related ? Since, after all, the main channels relying on special timers are Meetme, IAX and (maybe) MusicOnHold according to http://www.voip-info.org/wiki-Asterisk+timer Just want to be sure, since I still believe my mere demo playback issue likely has a different reason ... Thanks, Bruno. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
