Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the suggestion of someone on #asterisk, I cvsup'd * against digium and used that instead of ports, but that didn't seem to help either. FYI: When I said above "when I try to use the conference" I meant using two non-voip phones, specifically a cell phone and a land line. I'd dial the number for my asterix box which is in itself a b channel on a PRI answered by a T100P on a friend's * box and sent via IAX over to my * box. Not sure if that matters, but I figure I'd mention it anyway. Anyone have any ideas here? # meetme.conf [rooms] conf => 97531,24680 # extensions.conf [conf] exten => 1,1,Answer exten => 1,2,Wait(1) exten => 1,3,Authenticate(5447847) exten => 1,4,MeetMe(97531,Mas,24680) exten => 1,5,Playback(vm-goodbye) exten => 1,6,Hangup() exten => 2,1,MeetMe(97531,Ms,24680) [jlixfeld@trek://~ ]$ kldstat Id Refs Address Size Name 1 5 0xc0400000 5e16d8 kernel 2 4 0xc231e000 2f000 zaptel.ko 3 1 0xc234f000 6000 wcfxo.ko 4 1 0xc2355000 a000 wcfxs.ko 5 1 0xc235f000 2000 ztdummy.ko [jlixfeld@trek://~ ]$
Did you ever have any luck with improving the MeetMe performance? We're running into the same problems.... Andy -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Lixfeld Sent: Thursday, December 09, 2004 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Horrible MeetMe performance Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the suggestion of someone on #asterisk, I cvsup'd * against digium and used that instead of ports, but that didn't seem to help either. FYI: When I said above "when I try to use the conference" I meant using two non-voip phones, specifically a cell phone and a land line. I'd dial the number for my asterix box which is in itself a b channel on a PRI answered by a T100P on a friend's * box and sent via IAX over to my * box. Not sure if that matters, but I figure I'd mention it anyway. Anyone have any ideas here? # meetme.conf [rooms] conf => 97531,24680 # extensions.conf [conf] exten => 1,1,Answer exten => 1,2,Wait(1) exten => 1,3,Authenticate(5447847) exten => 1,4,MeetMe(97531,Mas,24680) exten => 1,5,Playback(vm-goodbye) exten => 1,6,Hangup() exten => 2,1,MeetMe(97531,Ms,24680) [jlixfeld@trek://~ ]$ kldstat Id Refs Address Size Name 1 5 0xc0400000 5e16d8 kernel 2 4 0xc231e000 2f000 zaptel.ko 3 1 0xc234f000 6000 wcfxo.ko 4 1 0xc2355000 a000 wcfxs.ko 5 1 0xc235f000 2000 ztdummy.ko [jlixfeld@trek://~ ]$ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Make sure that if you're using anything other than zaptel hardware, it is running uLaw as the codec. Anything else will produce ever increasing delays. My setup has all of our VoIP lines coming into my main box, and then I have a separate box running asterisk only for meetme with an iax2 trunk between the two running uLaw. It seems to work fairly well. Dan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andy Rosen Sent: Sunday, June 26, 2005 2:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Horrible MeetMe performance Did you ever have any luck with improving the MeetMe performance? We're running into the same problems.... Andy -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Lixfeld Sent: Thursday, December 09, 2004 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Horrible MeetMe performance Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the suggestion of someone on #asterisk, I cvsup'd * against digium and used that instead of ports, but that didn't seem to help either. FYI: When I said above "when I try to use the conference" I meant using two non-voip phones, specifically a cell phone and a land line. I'd dial the number for my asterix box which is in itself a b channel on a PRI answered by a T100P on a friend's * box and sent via IAX over to my * box. Not sure if that matters, but I figure I'd mention it anyway. Anyone have any ideas here? # meetme.conf [rooms] conf => 97531,24680 # extensions.conf [conf] exten => 1,1,Answer exten => 1,2,Wait(1) exten => 1,3,Authenticate(5447847) exten => 1,4,MeetMe(97531,Mas,24680) exten => 1,5,Playback(vm-goodbye) exten => 1,6,Hangup() exten => 2,1,MeetMe(97531,Ms,24680) [jlixfeld@trek://~ ]$ kldstat Id Refs Address Size Name 1 5 0xc0400000 5e16d8 kernel 2 4 0xc231e000 2f000 zaptel.ko 3 1 0xc234f000 6000 wcfxo.ko 4 1 0xc2355000 a000 wcfxs.ko 5 1 0xc235f000 2000 ztdummy.ko [jlixfeld@trek://~ ]$ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I read somewhere, probably the Wiki, that the core of MeetMe uses uLaw. So when sound comes in, it is transcoded to uLaw first, then mixed with all of the other audio at that moment, then sent out again. At that point, it is transcoded again to the original format. So, if everything is in uLaw, you bypass 2 transcoding processes. And if you take into account both of those transcoding processes (with other codecs), they add 10 or so milliseconds each time which results in ever increasing delays. If you are using all uLaw connections, try changing one of them to iLBC and try another conference. I noticed 10 second delays after 5 minutes of conference. Let me know if you find anything else out about this. Dan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tony Mountifield Sent: Sunday, June 26, 2005 5:35 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Horrible MeetMe performance In article <9FEE3E56D424E243B7CA76B875B07326045F6B@sbserver.ABBCO.inner>, Dan Morin <DMorin@ABBCOInc.com> wrote:> Make sure that if you're using anything other than zaptel hardware, it > is running uLaw as the codec. Anything else will produce ever > increasing delays.Hey, now that's a snippet of information I hadn't seen before! All my work with SIP and MeetMe is using aLaw, since I'm in the UK. Do you know why it causes ever increasing delays? I would have thought that a transcoding would just introduce a contstant (small) delay, not an accumulating one. So if you're right, then it ought to be fixable, once the mechanism is understood. Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
> In my experience, several seconds of delay becomes apparent over timewhen> using an internal clock source. Seems its a clocking/timer issue.Yes. Meetme can have horrible issues with timing. This _has_ been fixed. If you download the CVS Zaptel drivers, use a 2.6 kernel, you can use RTC support in ztdummy. Change the '#if 0' to '#if 1'. This is documented at http://www.aussievoip.com.au/wiki-AMP-Zaptel --Rob
> What's wrong with the standard 2.6 ztdummy?It doesn't use RTC. I'm assuming you mean '1.0.8' as 'standard'.> How does HEAD zaptel interact with 1.0 asterisk?Shouldn't cause any problems. --Rob
> If you have Zaptel cards, does setting the build to USE_RTC use that > timing source in preference to the Zaptel card interrupts?If you have a zaptel card, you shouldn't be loading ztdummy. So, therefore, no problems! --Rob