[asterisk-users] dimensioning
I have been trying to get a feel for scaling or dimensioning using asterisk 11. if I desire to use something like a dell r320, hardware RAID, 2G E5-2420, 4G RAM and only SIP trunking using gsm (least bandwidth and no transcoding) how many calls "out" can I expect to make at one time and asterisk still be OK and responsive? Thanks, Jerry -- _ -- 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] dimensioning
On 28/01/14 15:01, Jerry Geis wrote: I have been trying to get a feel for scaling or dimensioning using asterisk 11. if I desire to use something like a dell r320, hardware RAID, 2G E5-2420, 4G RAM and only SIP trunking using gsm (least bandwidth and no transcoding) how many calls "out" can I expect to make at one time and asterisk still be OK and responsive? Thanks, Jerry We have a quad 'Intel(R) Xeon(R) CPU E5-1410 0 @ 2.80GHz' that will copy with about 800 concurrent calls with a lot of AGI stuff the log call start and end and about 5-10% of the calls being recorded. All using g711alaw. All calls come in and go out via sip. -- _ -- 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] Dimensioning
On Thu, 17 Apr 2014, Jerry Geis wrote:
I was thinking transcoding was through PRI card - not gsm to ulaw. :)
You can convert the GSM files to ULAW using sox. I tend to transcode
everything to WAV (PCM not that funky 'GSM in WAV') because it is
relatively cheap (CPU cycles) to transcode from WAV to ULAW and everything
else in the world understands WAV just fine. If you really need to squeeze
out every last cycle, you can schedule a script to transcode WAVs to ULAWs
as needed.
So if all I am doing is originating calls, and using playback() in the
dialplan - then a system() call on completion I can expect upwards or
3000 concurrent calls?
Based on my unsubstantiated testing on my hosts, that seems like a
reasonable conclusion.
What do you do in the program executed by system()?
How do you actually test to make sure without having 3000 users to call.
Crowdsourcing?
No, it's really pretty simple.
On the 'source' host, I have a call file:
# sample-call-file
channel:sip/test@target
application:playback
data:/tmp/total
# (end of sample-call-file)
And a shell script to create the call files:
# create-calls.sh
cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
# (end of create-calls.sh)
Then, on the 'target' host I have a dialplan snippet:
[public]
exten = test,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = test,n, set(GROUP()=TEST)
exten = test,n, set(ROOM=0${GROUP_COUNT()})
exten = test,n, meetme(${ROOM:-2}, cd)
; exten = test,n, confbridge(${ROOM:-2})
exten = test,n, hangup()
Then, on the 'source' host, I can create calls with this command:
./create-calls.sh
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Dimensioning
Steve Edwards wrote: On Thu, 17 Apr 2014, Jerry Geis wrote: I was thinking transcoding was through PRI card - not gsm to ulaw. :) You can convert the GSM files to ULAW using sox. I tend to transcode everything to WAV (PCM not that funky 'GSM in WAV') because it is relatively cheap (CPU cycles) to transcode from WAV to ULAW and everything else in the world understands WAV just fine. If you really need to squeeze out every last cycle, you can schedule a script to transcode WAVs to ULAWs as needed. So if all I am doing is originating calls, and using playback() in the dialplan - then a system() call on completion I can expect upwards or 3000 concurrent calls? Based on my unsubstantiated testing on my hosts, that seems like a reasonable conclusion. I'll just chime in slightly here. At that number of calls you've got many factors going on - and they aren't just Asterisk. If playing back files you've got disk I/O going on which can slow stuff down (unless the system caches it enough or you throw them into a ramdisk yourself). You are also sending and receiving a *ton* of small packets. This can make network equipment and NICs unhappy. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.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] Dimensioning asterisk 11
I will be using a dell R320 Xeon E5-2420 2G and 4G RAM. also using a SIP trunk with ulaw/alaw codec. How many calls could I expect to make at the same time? no transcoding or anything. Just call a number and play a gsm file. Thanks, Jerry -- _ -- 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] Dimensioning asterisk 11
On Thu, 17 Apr 2014, Jerry Geis wrote: I will be using a dell R320 Xeon E5-2420 2G and 4G RAM.also using a SIP trunk with ulaw/alaw codec. no transcoding or anything. Just call a number and play a gsm file. How will you do ulaw <-> gsm without transcoding? How many calls could I expect to make at the same time? A whole bunch? It's hard to give any specifics without the same hardware and workload. Here's a datapoint to consider -- testing an HP ProLiant DL320e Gen8 v2 E3-1240v3 8GB. 9300 passmarks vs your 7300 passmarks. (And only $880 from Newegg.) 2 hosts, 1 originating calls, 1 running a simple dialplan, but similar to the expected production dialplan. 500 'participants' - 100 meetme conferences with 5 calls in each. 3000 'participants' - 100 confbridge conferences with 30 calls in each. Meetme() is still a 'single thread' application so you're done when you max out 1 CPU core. 500 calls was my goal, so that's where testing stopped. The hosts aren't in production yet, so I don't know if my testing experience will match production experience. I would expect playback() (without transcoding) to be significantly less CPU hungry than meetme() or confbridge(). -- 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
[asterisk-users] Dimensioning a 50 sip phone installation
Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one fonebridge (TDMoE) I was thinking of an HP DL140 with two 250gig sata disks and one 3.8Xeon CPU with 2gig RAM. Also, does a fonebridge setup suffers from the fact that 1.4 has no PRI/R2 support (as said in a previous post by someone else). -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dimensioning a telephony system based on openser!
Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work with this requisites: - 1 users; - 100 VoIP to VoIP calls simultaneously capacity; - 30 VoIP to PSTN calls simultaneously capacity; Can anyone point me some ideas of how can i design such a system (how many servers, how to distribute the services among them, etc.). I have this prototype mounted with VMWare, so i think that even making tests with sipp aren't going to be reliable. Thanks in advance, Nuno ___ -- 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] Dimensioning a 50 sip phone installation
Hi Erick - Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one fonebridge (TDMoE) I was thinking of an HP DL140 with two 250gig sata disks and one 3.8Xeon CPU with 2gig RAM. You'll do absolutely fine with this setup. I have an office that has about the same amount of phones and traffic as this (all using g711), but probably with quite a bit more conferencing. It runs on a Xeon 2.8ghz, 1GB Ram, 2 73 GB SCSI Raid 1. Also, does a fonebridge setup suffers from the fact that 1.4 has no PRI/R2 support (as said in a previous post by someone else). I've been considering buying one of these, but don't have one yet. If anyone can comment, I'd like an answer, too. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dimensioning a 50 sip phone installation
On 1/4/07, Noah Miller <[EMAIL PROTECTED]> wrote: Hi Erick - > Some help with dimensioning the server will be gladly accepted. > > -50 sip phones (g729) or g711(to avoid transcoding) in LAN > -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN > -Some sporadic conferencing with no more than 2 sip phones and maybe 2 > or 3 calls coming from the E1 for a total of 5 people in a conference. > > The asterisk server will get an E1(pri) via one fonebridge (TDMoE) > > I was thinking of an HP DL140 with two 250gig sata disks and one > 3.8Xeon CPU with 2gig RAM. You'll do absolutely fine with this setup. I have an office that has about the same amount of phones and traffic as this (all using g711), but probably with quite a bit more conferencing. It runs on a Xeon 2.8ghz, 1GB Ram, 2 73 GB SCSI Raid 1. > Also, does a fonebridge setup suffers from the fact that 1.4 has no > PRI/R2 support (as said in a previous post by someone else). I've been considering buying one of these, but don't have one yet. If anyone can comment, I'd like an answer, too. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks Noah, you have been very helpful. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dimensioning a 50 sip phone installation
I was thinking of an HP DL140 with two 250gig sata disks and one 3.8Xeon CPU with 2gig RAM. Should be plenty if not an overkill. One of our setups: 20 phones, 8 outgoing/incoming SIP trunks, MeetMe conferencing with ztdummy and no Zap hardware. IVR/voice mail/MOH/Recordings/etc. Runs on a single PIII-600, 256 MB RAM. CentOS 4.4 with a stock 2.6.9-42 kernel. Asterisk 1.2.5, in production for 1.5+ years. CPU usage about 2% per call. Quite reliable (hence not upgraded). This is a g711 only setup with no transcoding. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dimensioning a 50 sip phone installation
what if I go with full g711-no transcoding? remember that I will have an E1 coming in, so my usage can be up to 30 channels at once. if that is an overkill machine config, and for obvious reasons I cant use old hardware, what are your suggestions? thanks, On 1/5/07, Luki <[EMAIL PROTECTED]> wrote: > I was thinking of an HP DL140 with two 250gig sata disks and one > 3.8Xeon CPU with 2gig RAM. Should be plenty if not an overkill. One of our setups: 20 phones, 8 outgoing/incoming SIP trunks, MeetMe conferencing with ztdummy and no Zap hardware. IVR/voice mail/MOH/Recordings/etc. Runs on a single PIII-600, 256 MB RAM. CentOS 4.4 with a stock 2.6.9-42 kernel. Asterisk 1.2.5, in production for 1.5+ years. CPU usage about 2% per call. Quite reliable (hence not upgraded). This is a g711 only setup with no transcoding. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dimensioning a 50 sip phone installation
Erick Perez wrote: what if I go with full g711-no transcoding? remember that I will have an E1 coming in, so my usage can be up to 30 channels at once. if that is an overkill machine config, and for obvious reasons I cant use old hardware, what are your suggestions? I would suggest you go for a box that has redundant PSU. Most 1U boxes can't support redundant PSUs. IMHO, a 2U industrial PC with a single dual-core Pentium Dxxx 2.8GHz+ (or Xeon 3xxx) with hotswap RAID-1 HDD and PSU would be more than enough. I generally prefer 2U over 1U, because it's easier to cool and there's space to accommodate PCI cards of various sizes. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dimensioning a 50 sip phone installation
The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx. Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw phones +voicemail and *no* call recording? On 1/5/07, Leo Ann Boon <[EMAIL PROTECTED]> wrote: Erick Perez wrote: > what if I go with full g711-no transcoding? > remember that I will have an E1 coming in, so my usage can be up to 30 > channels at once. > if that is an overkill machine config, and for obvious reasons I cant > use old hardware, what are your suggestions? I would suggest you go for a box that has redundant PSU. Most 1U boxes can't support redundant PSUs. IMHO, a 2U industrial PC with a single dual-core Pentium Dxxx 2.8GHz+ (or Xeon 3xxx) with hotswap RAID-1 HDD and PSU would be more than enough. I generally prefer 2U over 1U, because it's easier to cool and there's space to accommodate PCI cards of various sizes. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dimensioning a 50 sip phone installation
On 7 Jan 2007, at 07:28, Erick Perez wrote: The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx. Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw phones +voicemail and *no* call recording? Yes. I've got the 1ghz version and it is fine (even doing 5 channels of alaw -> g729a) BUT I'm not entirely happy with these motherboards. I've just had 2 (out of a total of 5!) die. In both cases the capacitors were faulty. The machines were >1 year old and out of warranty, so I couldn't get replacements foc despite the fact that the caps were clearly sub-standard. When I replace these systems I won't be buying VIA EPIA again. If you do buy them, make sure you spec a CD drive in the package. I didn't and OS rebuilds are a pain to do via a USB DVD drive. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dimensioning a 50 sip phone installation
Erick Perez wrote: The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx. Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw phones +voicemail and *no* call recording? Make sure there're no interrupt sharing issues. My old EPIA 1GHz with 2 LAN and 6 USB, had no IO-APIC. It took lots of trial and error to make sure the digium card was not sharing an interrupt. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dimensioning: Where is the CPU vs Asterisk load table
Hi, is there any good calculator/table/reference about proper dimensioning? I read the wiki and they basically say "xx users run fine in yy hardware" http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning. SO far I read that: -Run up to 4 E1s per CPU (which one? an i386 or a dual core? -it is very CPU intensive to do transcoding. Try to minimize it. -you can help the CPU by using DSP-based boards or optimized boards. -Lots of RAM is good too. (like 512MB or 8 GB?). -A Front Side Bus of 800mhz is also good. -disable HT on Intel CPUs. -Use Ram disk to avoid some I/O bottlenecks specially on voicemail (hence, deploy more RAM). -two single core CPUs better than one dual core CPUs??. -And the most important I read was: Keep load under 5 in single CPUs and 10 in dual CPUs (didn't mention dual cores in the article). Im not sure If Im asking properly, basically in this asterisk-heavy-load-learning stage, I want to know how to calculate computer needs based on customer needs. (i've only done 5 to 50 extens, and up to 2 E1) And yes, the need comes from a potential customer that is a Hotel with 450 extensions (rooms) and 125 more ext. (employees) making calls internally, outbound, inboud, voicemail,fax,cell phones, etc. So far, shame on me, I have no idea where to start in terms of equipment. Or I can go out there and buy a 20k machine(s) to run 4or5 E1sIt will run, but I will never learn why. thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dimensioning: Where is the CPU vs Asterisk load table
On Thu, Jan 12, 2006 at 03:05:15PM -0500, Erick Perez wrote: > Hi, is there any good calculator/table/reference about proper dimensioning? > I read the wiki and they basically say "xx users run fine in yy hardware" > http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning. > > SO far I read that: > -Run up to 4 E1s per CPU (which one? an i386 or a dual core? This is just too general to be true. A dual-CPU PIII monster of a number of years ago is pathertic cmpared to a single 3GHz P4. > -it is very CPU intensive to do transcoding. Try to minimize it. > -you can help the CPU by using DSP-based boards or optimized boards. Up to a point. Normally it will be cheaper to just throw in more CPU power. > -Lots of RAM is good too. (like 512MB or 8 GB?). Depending on the task. Lots of RAM helps if you have much IO. > -A Front Side Bus of 800mhz is also good. This follows the "get a fast machine" rule. > -disable HT on Intel CPUs. > -Use Ram disk to avoid some I/O bottlenecks specially on voicemail > (hence, deploy more RAM). > -two single core CPUs better than one dual core CPUs??. > -And the most important I read was: Keep load under 5 in single CPUs > and 10 in dual CPUs (didn't mention dual cores in the article). > > Im not sure If Im asking properly, basically in this > asterisk-heavy-load-learning stage, I want to know how to calculate > computer needs based on customer needs. > (i've only done 5 to 50 extens, and up to 2 E1) That can probably be easily handled by the typical computer you buy on on any shop. Speed is not your main issue. Make sure the system is relieble. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
