Re: [asterisk-users] Internal timing under load is critical ?
> As a matter of curiosity, what do people use these voice broadcasting > solutions for? > > I'm genuinely struggling to think of (legal) reasons why you'd want to > broadcast 1000+ simultaneous calls. Perhaps I'm just not being > imaginative enough... :-) > > Kind regards, > > Chris For a big PSTN operator (+10 million subscribers) there are many applications for voice broadcasting,just one sample if they can get postpaid phone bills one week sooner or if they can reduce number of unpaid bills by voice notices they compensate all expenses. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Internal timing under load is critical ?
On 07/31/2014 09:51 AM, Chris Bagnall wrote: On 30/7/14 10:08 am, babak wrote: I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. As a matter of curiosity, what do people use these voice broadcasting solutions for? I'm genuinely struggling to think of (legal) reasons why you'd want to broadcast 1000+ simultaneous calls. Perhaps I'm just not being imaginative enough... :-) Kind regards, Chris Community and emergency notification systems are the first uses to come to mind. That said, "Jane from Card Services" is probably much more common. -- Daniel Taylor VP OperationsVocal Laboratories, Inc. dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Internal timing under load is critical ?
On 30/7/14 10:08 am, babak wrote: I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. As a matter of curiosity, what do people use these voice broadcasting solutions for? I'm genuinely struggling to think of (legal) reasons why you'd want to broadcast 1000+ simultaneous calls. Perhaps I'm just not being imaginative enough... :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Internal timing under load is critical ?
On Wed, Jul 30, 2014 at 8:33 PM, Steve Edwards wrote: > which timing module you are using: res_timing_timerfd.so or > res_timing_kqueue.so or res_timing_dahdi.sores_timing_pthread.so > > I used res_timing_timerfd.so. I'm finally making the leap from 1.2 to the > current decade :) I read somewhere that this was the timer to use and it > seems to be working fine for me. > Welcome to the current decade! Glad to hear Asterisk 11 is working out okay for you so far. 1.2 to 11 is a big leap. FWIW, the Asterisk wiki has a run down on timing interfaces and which ones are preferred: https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces For those who want the cliff's notes version, it is: * res_timing_timerfd * res_timing_kqueue (where available) * res_timing_dahdi * res_timing_pthread In particular, res_timing_pthread should only be used as a last resort. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Internal timing under load is critical ?
Please don't top post. Please keep the thread only on the list. On Thursday, July 31, 2014 12:16 AM, Steve Edwards wrote: I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2. 1,300 calls with no audio issues. On Wed, 30 Jul 2014, babak wrote: 1300 calls include playback voices ? The test scenario was for the first server to originate calls (via call files) to the second server and then 'playback()' a long file. The second server would answer the call and then 'playback()' a long file. Audio was flowing in each direction. Bandwidth was observed using 'iftop' as being in the 70mb to 80mb range in each direction (if I remember correctly). I placed calls from a handset to confirm audio quality. which timing module you are using: res_timing_timerfd.so or res_timing_kqueue.so or res_timing_dahdi.sores_timing_pthread.so I used res_timing_timerfd.so. I'm finally making the leap from 1.2 to the current decade :) I read somewhere that this was the timer to use and it seems to be working fine for me. I don't think the cores got much over 20% to 30% busy. Various failures were observed on the console from running out of file descriptors. This was on a stock CentOS 6.5 install with no tweaks to bump up the max file descriptors. The client only asked for 500 simultaneous calls so no further testing was done. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Internal timing under load is critical ?
Thank you for replys 1300 calls include playback voices ? which timing module you are using: res_timing_timerfd.so orres_timing_kqueue.so or res_timing_dahdi.sores_timing_pthread.so Regards On Thursday, July 31, 2014 12:16 AM, Steve Edwards wrote: On Wed, 30 Jul 2014, babak wrote: > According to some recommendations like http://osdial.org/howto/ > "Internal timing is very critical with Asterisk when it is under load" > and we must use DAHDI hardware or "USB Voice Synch Tool" > http://www.sangoma.com/accessories/specialty-tools/ But according to my > understanding of wiki > https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces It seems it > is not necessary now. Please tell me your opinions. I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2. 1,300 calls with no audio issues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Internal timing under load is critical ?
On Wed, 30 Jul 2014, babak wrote: According to some recommendations like http://osdial.org/howto/ "Internal timing is very critical with Asterisk when it is under load" and we must use DAHDI hardware or "USB Voice Synch Tool" http://www.sangoma.com/accessories/specialty-tools/ But according to my understanding of wiki https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces It seems it is not necessary now. Please tell me your opinions. I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2. 1,300 calls with no audio issues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Internal timing under load is critical ?
If you were running on Asterisk 1.4, a Zaptel or Dahdi timing source (including the Sangoma USB device) was necessary to avoid sometimes unreliable timing from the "dummy" interface. For modern releases (1.6, 1.8, 11, 12, etc) this isn't necessary for most systems. However, you may have better results with such a large number of calls by using a hardware timing source. The difference will vary between different systems and loads -- I recommend testing it on your own platform. Note that changing to a different model with a different motherboard or even just a different chipset can result in a difference in timing accuracy. -- so your best option is to try it both ways under load to see if you see a benefit, and re-test should you change the platform, such as using a different motherboard. On Wed, Jul 30, 2014 at 4:08 AM, babak wrote: > Hi > I am evaluating some voice broadcasting solutions based on Asterisks for > more than 1000 simultaneous calls. > Connection to Asterisk all are based on SIP and SIP Trunks so no DAHDI > hardware is required. > According to some recommendations like http://osdial.org/howto/ > "Internal timing is very critical with Asterisk when it is under load" > and we must use DAHDI hardware or "USB Voice Synch Tool" > http://www.sangoma.com/accessories/specialty-tools/ > But according to my understanding of wiki > https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces > It seems it is not necessary now. > Please tell me your opinions. > > Regards > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Internal timing under load is critical ?
Hi I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. Connection to Asterisk all are based on SIP and SIP Trunks so no DAHDI hardware is required. According to some recommendations like http://osdial.org/howto/ "Internal timing is very critical with Asterisk when it is under load" and we must use DAHDI hardware or "USB Voice Synch Tool" http://www.sangoma.com/accessories/specialty-tools/ But according to my understanding of wiki https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces It seems it is not necessary now. Please tell me your opinions. Regards-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
