2012 VoIP With Virtualization
Hi FS Users, What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA. If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... Thanks for your inputs Chris
basically it depends ... on MANY things.. a) how much you care about your voice quality b) how many concurrent calls c) do you modify the RTP stream or just pass it through. basically if your real picky or have more than a handful of concurrent calls.. then dont do it. if your using conferences etc... dont do it. that being said, you can get away with a lot if you dont mind the occasional blip and chirp in your media.. Thing to not though... above is for virtualization ... ( ESX, Xen, etc ) Jails like OpenVZ, BSD Jails, etc work well and provide many of the benefits found in virtualization. Jay On 15 May 2012 19:12, Chris Mylonas <chris@opencsta.org> wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely Jay
Beauty - thanks Jay - I'll check out what jails can do for me. I've just compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb not being available. In my world, voice is precious ;) No virtualisation continues to be my modus operandi for now then :) Cheers Chris On 15/05/2012, at 8:32 PM, jay binks wrote:
basically it depends ... on MANY things..
a) how much you care about your voice quality b) how many concurrent calls c) do you modify the RTP stream or just pass it through.
basically if your real picky or have more than a handful of concurrent calls.. then dont do it. if your using conferences etc... dont do it.
that being said, you can get away with a lot if you dont mind the occasional blip and chirp in your media..
Thing to not though... above is for virtualization ... ( ESX, Xen, etc )
Jails like OpenVZ, BSD Jails, etc work well and provide many of the benefits found in virtualization.
Jay
On 15 May 2012 19:12, Chris Mylonas <chris@opencsta.org> wrote: Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely
Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas <chris@opencsta.org> wrote:
Beauty - thanks Jay - I'll check out what jails can do for me. I've just compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb not being available.
Chris, obviously do not forget for mod_gsmopen to "export" the serial devices from the "host" to the "guest", eg: from the bare metal to the containers. I use that daily, works perfect. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
I mean, OpenVZ works perfect for mod_gsmopen, that I was meaning. Also, there is consensus from the core developers that OpenVZ is the virtualization platform works with FreeSWITCH (mod_gsmopen and mod_skypopen included) without particular tweak. That's because OpenVZ is a kind of jail, so applications in guests have direct access to the IRQs of the one only kernel running on the bare metal (as opposed to Xen, VMWare, KVM, etc that runs many concurrent kernels on software simulated hardware machines). -giovanni On Wed, May 16, 2012 at 5:55 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas <chris@opencsta.org> wrote:
Beauty - thanks Jay - I'll check out what jails can do for me. I've just compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb not being available.
Chris, obviously do not forget for mod_gsmopen to "export" the serial devices from the "host" to the "guest", eg: from the bare metal to the containers. I use that daily, works perfect.
-giovanni
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
I'm in the process of setting up proxmox.com's KVM & OpenVZ on an old core 2 duo - so that will test the pants off the resources :) Thanks for the heads up. On 17/05/2012, at 2:01 AM, Giovanni Maruzzelli wrote:
I mean, OpenVZ works perfect for mod_gsmopen, that I was meaning.
Also, there is consensus from the core developers that OpenVZ is the virtualization platform works with FreeSWITCH (mod_gsmopen and mod_skypopen included) without particular tweak.
That's because OpenVZ is a kind of jail, so applications in guests have direct access to the IRQs of the one only kernel running on the bare metal (as opposed to Xen, VMWare, KVM, etc that runs many concurrent kernels on software simulated hardware machines).
-giovanni
On Wed, May 16, 2012 at 5:55 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas <chris@opencsta.org> wrote:
Beauty - thanks Jay - I'll check out what jails can do for me. I've just compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb not being available.
Chris, obviously do not forget for mod_gsmopen to "export" the serial devices from the "host" to the "guest", eg: from the bare metal to the containers. I use that daily, works perfect.
-giovanni
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
please let us know all you will know :) On Wed, May 16, 2012 at 6:14 PM, Chris Mylonas <chris@opencsta.org> wrote:
I'm in the process of setting up proxmox.com's KVM & OpenVZ on an old core 2 duo - so that will test the pants off the resources :) Thanks for the heads up.
On 17/05/2012, at 2:01 AM, Giovanni Maruzzelli wrote:
I mean, OpenVZ works perfect for mod_gsmopen, that I was meaning.
Also, there is consensus from the core developers that OpenVZ is the virtualization platform works with FreeSWITCH (mod_gsmopen and mod_skypopen included) without particular tweak.
That's because OpenVZ is a kind of jail, so applications in guests have direct access to the IRQs of the one only kernel running on the bare metal (as opposed to Xen, VMWare, KVM, etc that runs many concurrent kernels on software simulated hardware machines).
-giovanni
On Wed, May 16, 2012 at 5:55 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas <chris@opencsta.org> wrote:
Beauty - thanks Jay - I'll check out what jails can do for me. I've just compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb not being available.
Chris, obviously do not forget for mod_gsmopen to "export" the serial devices from the "host" to the "guest", eg: from the bare metal to the containers. I use that daily, works perfect.
-giovanni
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Sure... when I get stuff done. In my best efforts not to stuff up what I have, I resized my RAID from 2.8TB to 790GB accidentally. I wanted it set to 2000GB, not reduced by 2000GB!!! No damage luckily :) Here are proxmox's features: http://proxmox.com/products/proxmox-ve/features and they are impressive. The same host can do both KVM and OpenVZ - great for testing I guess. Parts are AGPL licensed, so if you were to offer any network services to clients (i.e. the REST API) - you know what that means... Nothing stopping you from making a front end for it that interacts with that web service though. The GUI is very polished from my couple of hours of waiting for my poor-man's-SAN to resize :) I won't have solid info on it for a while (i'm going up north for a few weeks so my output is greatly reduced), I'll blog about most of my stuff, including the telephony load testing. Cheers Chris On 17/05/2012, at 2:17 AM, Giovanni Maruzzelli wrote:
please let us know all you will know :)
On Wed, May 16, 2012 at 6:14 PM, Chris Mylonas <chris@opencsta.org> wrote:
I'm in the process of setting up proxmox.com's KVM & OpenVZ on an old core 2 duo - so that will test the pants off the resources :) Thanks for the heads up.
On 17/05/2012, at 2:01 AM, Giovanni Maruzzelli wrote:
I mean, OpenVZ works perfect for mod_gsmopen, that I was meaning.
Also, there is consensus from the core developers that OpenVZ is the virtualization platform works with FreeSWITCH (mod_gsmopen and mod_skypopen included) without particular tweak.
That's because OpenVZ is a kind of jail, so applications in guests have direct access to the IRQs of the one only kernel running on the bare metal (as opposed to Xen, VMWare, KVM, etc that runs many concurrent kernels on software simulated hardware machines).
-giovanni
On Wed, May 16, 2012 at 5:55 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas <chris@opencsta.org> wrote:
Beauty - thanks Jay - I'll check out what jails can do for me. I've just compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb not being available.
Chris, obviously do not forget for mod_gsmopen to "export" the serial devices from the "host" to the "guest", eg: from the bare metal to the containers. I use that daily, works perfect.
-giovanni
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks G - the virtualization is a different set of testing. My iSCSI target (file server) has FS on it as well with the USB dongle (btw, i already had DEBUG=0, so will review). The virtualization is testing for queuing - have SIPp agents answer calls and echo the media. I only have a 100Mbit switch though. Thanks for the thought though :) On 17/05/2012, at 1:55 AM, Giovanni Maruzzelli wrote:
On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas <chris@opencsta.org> wrote:
Beauty - thanks Jay - I'll check out what jails can do for me. I've just compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb not being available.
Chris, obviously do not forget for mod_gsmopen to "export" the serial devices from the "host" to the "guest", eg: from the bare metal to the containers. I use that daily, works perfect.
-giovanni
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, Chris That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic? - ¿How many calls have you handle in the real world? - ¿Transcoding? - ¿Have you tested your VM? Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!! Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware: http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.ht... http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf On 5/15/2012 4:42 AM, Chris Mylonas wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12
-- Atentamente, *Dario García* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve
Sent from mobile, sorry for the typos.... Em 15/05/2012 09:42, "Saugort Dario Garcia Tovar" <dgarcia@anew.com.ve> escreveu:
Hi, Chris
That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic? - ¿How many calls have you handle in the real world? - ¿Transcoding? - ¿Have you tested your VM?
Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!!
Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware:
http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.ht... http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf
On 5/15/2012 4:42 AM, Chris Mylonas wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12
-- Atentamente, *Dario García* Consultor.
CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
As said before, it depends heavily in your setup... We do virtualize FreeSWITCH and aside from a few glitches it has been good... We have approx 400 calls per VM. Later when I hit the office I can describe my setup in detail. Sent from mobile, sorry for the typos.... Em 15/05/2012 09:42, "Saugort Dario Garcia Tovar" <dgarcia@anew.com.ve> escreveu:
Hi, Chris
That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic? - ¿How many calls have you handle in the real world? - ¿Transcoding? - ¿Have you tested your VM?
Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!!
Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware:
http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.ht... http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf
On 5/15/2012 4:42 AM, Chris Mylonas wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12
-- Atentamente, *Dario García* Consultor.
CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
OK 400 calls is on the very high end of what I would have put on a VM. If you wouldn't mind explaining your VM set up that would be great. On 15/05/2012, at 10:54 PM, Wesley Akio wrote:
As said before, it depends heavily in your setup... We do virtualize FreeSWITCH and aside from a few glitches it has been good... We have approx 400 calls per VM. Later when I hit the office I can describe my setup in detail.
Sent from mobile, sorry for the typos....
Em 15/05/2012 09:42, "Saugort Dario Garcia Tovar" <dgarcia@anew.com.ve> escreveu: Hi, Chris
That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic? - ¿How many calls have you handle in the real world? - ¿Transcoding? - ¿Have you tested your VM?
Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!!
Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware: http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.ht... http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf
On 5/15/2012 4:42 AM, Chris Mylonas wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12
-- Atentamente, Dario García Consultor.
CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Sorry for the delay... For the setup in question: The hardware: IBM SystemX 3650. 8 core Xeons, E5*. 48GB RAM Virtualization is done by VMWare ESXi, mostly 4 or 5. Network is all Cisco. 3 VM Boxes that reach an average of 400 calls during working hours. FreesSWITCH there is very old(1.0.6). 4 CPU Cores 2GB 64 bit Centos 5.2 local dialplan XML no transcoding xml_cdr very stable. Servers have several virtual machines with a lot of different things running, FSs have priority on hardware resources... We have some other VM boxes(FS Sep 2011) with less load(270 calls avg) that do trancoding(25 calls avg g729<->g711), no quality issues but the FS clock drifts a lot(up to 10 minutes a day), VMware's fault -> http://www.vmware.com/files/pdf/Timekeeping-In-VirtualMachines.pdf fsctl sync_clock solves the problem but might ruin CDRs so BEWARE! There are several other VMs in quite a few hardware setups, but none of them has impressive call volumes... P.S.: Please don't shout at the age of the installs... they have served us well! :D. [ ]s, Wesley Akio TuntsCorp.com On Tue, May 15, 2012 at 10:08 AM, Chris Mylonas <chris@opencsta.org> wrote:
OK 400 calls is on the very high end of what I would have put on a VM. If you wouldn't mind explaining your VM set up that would be great.
On 15/05/2012, at 10:54 PM, Wesley Akio wrote:
As said before, it depends heavily in your setup... We do virtualize FreeSWITCH and aside from a few glitches it has been good... We have approx 400 calls per VM. Later when I hit the office I can describe my setup in detail.
Sent from mobile, sorry for the typos.... Em 15/05/2012 09:42, "Saugort Dario Garcia Tovar" <dgarcia@anew.com.ve> escreveu:
Hi, Chris
That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic? - ¿How many calls have you handle in the real world? - ¿Transcoding? - ¿Have you tested your VM?
Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!!
Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware:
http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.ht... http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf
On 5/15/2012 4:42 AM, Chris Mylonas wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12
-- Atentamente, *Dario García* Consultor.
CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
FreeSWITCH Version 1.2.0 (git-44fd0de 2012-05-14 02-04-36 +0200) on CentOS 6.2 Here is the scenario: Caller calls into dialplan, lua script gets called, their number gets parsed, does a db look up, then the caller is transferred to a recording then eventually hold music in dialplan. During the original callers session, script 2 gets called (see script 1). Script 2 executes lua script 3 on the origination. Speech is always detected on the session:answer(). I have even tried separate/new/independent sessions of the original session No go. The idea is to bridge the calls in a uuid bridge eventually and it works fine when it gets there, but I cannot trap when the call is answered under any scenario (no early media signaling and disposition is always 'ANSWER'), hence the ASR (tried all the execute_on_answer variants plus nolocal: yada yada, even tried it with python and javascript as well). I am cross-eyed by now so it is probably missing something stupid. I need the async behavior to keep the user on hold for as little time as possible. Any ideas are appreciated. TIA, Noel <script 1> caller_leg = session:getVariable("uuid") dealer = get_dealer_number(area_code) session:execute("export","store_name="..dealer["name"]) session:execute("export","store_id="..dealer["store_id"]) set_store_call(caller_leg, caller_num, tostring(dealer["store_number"])) api:executeString("luarun originate_dealer.lua " .. caller_leg .. " " .. dealer["store_number"] .. " " .. caller_num) session:execute("export", "store_number=" .. dealer["store_number"]) session:execute("export","caller_leg="..caller_leg) session:execute("export","caller_num="..caller_num) session:setHangupHook("session_hangup_hook") session:transfer("xxxxxx_park", "XML", "default") </script 1> <script 2> api = freeswitch.API() api:executeString("bgapi originate {caller_leg=".. argv[1] ..",origination_caller_id_number=".. argv[3] .."}sofia/gateway/provider/".. argv[2] .. " &lua(dealer.lua)") </script 2> <script 3> log("dealer...") caller_leg = session:getVariable("caller_leg") my_leg = session:getVariable("uuid") store_caller = get_caller(caller_leg) function onInput(s, type, obj) freeswitch.consoleLog("info", "Callback with type " .. type .. "\n"); if (type == "dtmf") then freeswitch.consoleLog("info", "DTMF Digit: " .. obj.digit .. "\n"); else if (type == "event") then local event = obj:getHeader("Speech-Type"); if (event == "begin-speaking") then freeswitch.consoleLog("info", "\n" .. obj:serialize() .. "\n"); -- Return break on begin-speaking events to stop playback of the fire or tts. call_answered=true return "break"; end if (event == "detected-speech") then freeswitch.consoleLog("info", "\n" .. obj:serialize() .. "\n"); if (obj:getBody()) then -- Pause speech detection (this is on auto but pausing it just in case) session:execute("detect_speech", "pause"); -- Parse the results from the event into the results table for later use. call_answered=true end return "break"; end end end end called_leg = session:getVariable("uuid") session:answer(); session:setInputCallback("onInput"); session:sleep(1000) session:execute("detect_speech", "pocketsphinx myapp default") while call_answered==false and tries <= 10 do session:sleep(2000) log("call answered is false...") tries = tries + 1 end if session:ready()==true then log(session:getVariable("endpoint_disposition")) digits = session:playAndGetDigits(1, 1, 1, 4000, '', '/usr/local/freeswitch/recordings/DealerGreeting.wav', '', '\\d+') if digits then if digits == "1" then log("In Dealer IVR and 1 was pressed...") api:executeString("uuid_bridge " ..called_leg .. " " .. caller_leg) else log("In Dealer IVR and 1 was NOT pressed...") api:executeString("uuid_bridge " ..called_leg .. " " .. caller_leg) end else n_session = freeswitch.Session() ses = freeswitch.Session("{effective_caller_id_number=".. caller_leg ..",ignore_early_media=true,originate_timeout=40}sofia/gateway/provider/2143 909113", n_session) called_leg = ses:getVariable("uuid") sess:waitForAnswer(sess) sess:answer() api:executeString("uuid_bridge " ..called_leg .. " " .. caller_leg) end end </script 3> 2012-05-15 17:01:09.748559 [DEBUG] switch_rtp.c:3253 Correct ip/port confirmed. INFO: cmd_ln.c(691): Parsing command line: \ -samprate 8000 \ -hmm /usr/local/freeswitch/grammar/model/communicator \ -jsgf /usr/local/freeswitch/grammar/xxxxxx.gram \ -lw 6.5 \ -dict /usr/local/freeswitch/grammar/default.dic \ -frate 50 \ -silprob 0.005 Current configuration: [NAME] [DEFLT] [VALUE] -agc none none -agcthresh 2.0 2.000000e+00 -alpha 0.97 9.700000e-01 -ascale 20.0 2.000000e+01 -aw 1 1 -backtrace no no -beam 1e-48 1.000000e-48 -bestpath yes yes -bestpathlw 9.5 9.500000e+00 -bghist no no -ceplen 13 13 -cmn current current -cmninit 8.0 8.0 -compallsen no no -debug 0 -dict /usr/local/freeswitch/grammar/default.dic -dictcase no no -dither no no -doublebw no no -ds 1 1 -fdict -feat 1s_c_d_dd 1s_c_d_dd -featparams -fillprob 1e-8 1.000000e-08 -frate 100 50 -fsg -fsgusealtpron yes yes -fsgusefiller yes yes -fwdflat yes yes -fwdflatbeam 1e-64 1.000000e-64 -fwdflatefwid 4 4 -fwdflatlw 8.5 8.500000e+00 -fwdflatsfwin 25 25 -fwdflatwbeam 7e-29 7.000000e-29 -fwdtree yes yes -hmm /usr/local/freeswitch/grammar/model/communicator -input_endian little little -jsgf /usr/local/freeswitch/grammar/xxxxxx.gram -kdmaxbbi -1 -1 -kdmaxdepth 0 0 -kdtree -latsize 5000 5000 -lda -ldadim 0 0 -lextreedump 0 0 -lifter 0 0 -lm -lmctl -lmname default default -logbase 1.0001 1.000100e+00 -logfn -logspec no no -lowerf 133.33334 1.333333e+02 -lpbeam 1e-40 1.000000e-40 -lponlybeam 7e-29 7.000000e-29 -lw 6.5 6.500000e+00 -maxhmmpf -1 -1 -maxnewoov 20 20 -maxwpf -1 -1 -mdef -mean -mfclogdir -min_endfr 0 0 -mixw -mixwfloor 0.0000001 1.000000e-07 -mllr -mmap yes yes -ncep 13 13 -nfft 512 512 -nfilt 40 40 -nwpen 1.0 1.000000e+00 -pbeam 1e-48 1.000000e-48 -pip 1.0 1.000000e+00 -pl_beam 1e-10 1.000000e-10 -pl_pbeam 1e-5 1.000000e-05 -pl_window 0 0 -rawlogdir -remove_dc no no -round_filters yes yes -samprate 16000 8.000000e+03 -seed -1 -1 -sendump -senlogdir -senmgau -silprob 0.005 5.000000e-03 -smoothspec no no -svspec -tmat -tmatfloor 0.0001 1.000000e-04 -topn 4 4 -topn_beam 0 0 -toprule -transform legacy legacy -unit_area yes yes -upperf 6855.4976 6.855498e+03 -usewdphones no no -uw 1.0 1.000000e+00 -var -varfloor 0.0001 1.000000e-04 -varnorm no no -verbose no no -warp_params -warp_type inverse_linear inverse_linear -wbeam 7e-29 7.000000e-29 -wip 0.65 6.500000e-01 -wlen 0.025625 2.562500e-02 EXECUTE sofia/xxxxxx/2143909113 detect_speech(pocketsphinx xxxxxx default) INFO: cmd_ln.c(691): Parsing command line: \ -alpha 0.97 \ -dither yes \ -doublebw no \ -nfilt 31 \ -ncep 13 \ -lowerf 200 \ -upperf 3500 \ -nfft 256 \ -wlen 0.0256 \ -transform legacy \ -feat s2_4x \ -agc none \ -cmn current \ -varnorm no Current configuration: [NAME] [DEFLT] [VALUE] -agc none none -agcthresh 2.0 2.000000e+00 -alpha 0.97 9.700000e-01 -ceplen 13 13 -cmn current current -cmninit 8.0 8.0 -dither no yes -doublebw no no -feat 1s_c_d_dd s2_4x -frate 100 50 -input_endian little little -lda -ldadim 0 0 -lifter 0 0 -logspec no no -lowerf 133.33334 2.000000e+02 -ncep 13 13 -nfft 512 256 -nfilt 40 31 -remove_dc no no -round_filters yes yes -samprate 16000 8.000000e+03 -seed -1 -1 -smoothspec no no -svspec -transform legacy legacy -unit_area yes yes -upperf 6855.4976 3.500000e+03 -varnorm no no -verbose no no -warp_params -warp_type inverse_linear inverse_linear -wlen 0.025625 2.560000e-02 INFO: acmod.c(242): Parsed model-specific feature parameters from /usr/local/freeswitch/grammar/model/communicator/feat.params INFO: fe_interface.c(289): You are using the internal mechanism to generate the seed. INFO: feat.c(684): Initializing feature stream to type: 's2_4x', ceplen=13, CMN='current', VARNORM='no', AGC='none' INFO: cmn.c(142): mean[0]= 12.00, mean[1..12]= 0.0 INFO: mdef.c(520): Reading model definition: /usr/local/freeswitch/grammar/model/communicator/mdef INFO: bin_mdef.c(173): Allocating 104160 * 8 bytes (813 KiB) for CD tree INFO: tmat.c(205): Reading HMM transition probability matrices: /usr/local/freeswitch/grammar/model/communicator/transition_matrices INFO: acmod.c(117): Attempting to use SCHMM computation module INFO: ms_gauden.c(198): Reading mixture gaussian parameter: /usr/local/freeswitch/grammar/model/communicator/means INFO: ms_gauden.c(292): 1 codebook, 4 feature, size: INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(294): 256x24 INFO: ms_gauden.c(294): 256x3 INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(198): Reading mixture gaussian parameter: /usr/local/freeswitch/grammar/model/communicator/variances INFO: ms_gauden.c(292): 1 codebook, 4 feature, size: INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(294): 256x24 INFO: ms_gauden.c(294): 256x3 INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(354): 59 variance values floored INFO: s2_semi_mgau.c(908): Loading senones from dump file /usr/local/freeswitch/grammar/model/communicator/sendump INFO: s2_semi_mgau.c(932): BEGIN FILE FORMAT DESCRIPTION INFO: s2_semi_mgau.c(995): Rows: 256, Columns: 6256 INFO: s2_semi_mgau.c(1027): Using memory-mapped I/O for senones INFO: s2_semi_mgau.c(1304): Maximum top-N: 4 Top-N beams: 0 0 0 0 INFO: dict.c(306): Allocating 137548 * 32 bytes (4298 KiB) for word entries INFO: dict.c(321): Reading main dictionary: /usr/local/freeswitch/grammar/default.dic INFO: dict.c(212): Allocated 1010 KiB for strings, 1664 KiB for phones INFO: dict.c(324): 133436 words read INFO: dict.c(330): Reading filler dictionary: /usr/local/freeswitch/grammar/model/communicator/noisedict INFO: dict.c(212): Allocated 0 KiB for strings, 0 KiB for phones INFO: dict.c(333): 17 words read INFO: dict2pid.c(396): Building PID tables for dictionary INFO: dict2pid.c(404): Allocating 51^3 * 2 bytes (259 KiB) for word-initial triphones INFO: dict2pid.c(131): Allocated 62832 bytes (61 KiB) for word-final triphones INFO: dict2pid.c(195): Allocated 62832 bytes (61 KiB) for single-phone word triphones INFO: fsg_search.c(145): FSG(beam: -1080, pbeam: -1080, wbeam: -634; wip: -26, pip: 0) INFO: jsgf.c(546): Defined rule: <xxxxxx.g00000> INFO: jsgf.c(546): Defined rule: <xxxxxx.yes> INFO: jsgf.c(546): Defined rule: <xxxxxx.g00002> INFO: jsgf.c(546): Defined rule: <xxxxxx.no> INFO: jsgf.c(546): Defined rule: <xxxxxx.g00004> INFO: jsgf.c(546): Defined rule: <xxxxxx.hello> INFO: jsgf.c(546): Defined rule: <xxxxxx.g00006> INFO: jsgf.c(546): Defined rule: PUBLIC <xxxxxx.results> INFO: fsg_model.c(213): Computing transitive closure for null transitions INFO: fsg_model.c(264): 87 null transitions added INFO: fsg_model.c(411): Adding silence transitions for <sil> to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++AE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++AH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++BACKGROUND++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++BREATH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++COUGH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++EH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++ER++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++LAUGH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++MM++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++MUMBLE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++OH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++SMACK++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UH_NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UM++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UM_NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_search.c(364): Added 1 alternate word transitions INFO: fsg_lextree.c(108): Allocated 2392 bytes (2 KiB) for left and right context phones INFO: fsg_lextree.c(251): 505 HMM nodes in lextree (496 leaves) INFO: fsg_lextree.c(253): Allocated 64640 bytes (63 KiB) for all lextree nodes INFO: fsg_lextree.c(256): Allocated 63488 bytes (62 KiB) for lextree leafnodes 2012-05-15 17:01:11.428551 [DEBUG] switch_core_media_bug.c:502 Attaching BUG to sofia/xxxxxx/2143909113 2012-05-15 17:01:11.448552 [DEBUG] switch_core_io.c:353 Setting BUG Codec PCMU:0 2012-05-15 17:01:13.448551 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) 2012-05-15 17:01:15.008548 [INFO] switch_cpp.cpp:1227 Callback with type event 2012-05-15 17:01:15.008548 [INFO] switch_cpp.cpp:1227 'Event-Name: DETECTED_SPEECH Core-UUID: 0a387574-9ee8-11e1-8fa1-43045b098f7a FreeSWITCH-Hostname: xxxxxx FreeSWITCH-Switchname: xxxxxx FreeSWITCH-IPv4: 199.15.96.29 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-05-15%2017%3A01%3A14 Event-Date-GMT: Wed,%2016%20May%202012%2000%3A01%3A14%20GMT Event-Date-Timestamp: 1337126474988540 Event-Calling-File: switch_ivr_async.c Event-Calling-Function: speech_thread Event-Calling-Line-Number: 3549 Event-Sequence: 1204 Speech-Type: begin-speaking ' 2012-05-15 17:01:15.008548 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) 2012-05-15 17:01:17.008548 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) INFO: cmn_prior.c(121): cmn_prior_update: from < 8.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 > INFO: cmn_prior.c(139): cmn_prior_update: to < 8.11 -0.00 0.40 -0.00 -0.39 -0.64 -0.79 -0.65 -0.51 -0.32 -0.08 0.12 0.11 > INFO: fsg_search.c(1030): 133 frames, 1846 HMMs (13/fr), 5417 senones (40/fr), 1099 history entries (8/fr) INFO: fsg_search.c(1407): Start node hello.0:14:96 INFO: fsg_search.c(1446): End node <sil>.95:105:132 (-2846) INFO: fsg_search.c(1662): lattice start node hello.0 end node <sil>.95 INFO: ps_lattice.c(1352): Normalizer P(O) = alpha(<sil>:95:132) = -886169 INFO: ps_lattice.c(1390): Joint P(O,S) = -886169 P(S|O) = 0 2012-05-15 17:01:17.648543 [DEBUG] mod_pocketsphinx.c:383 Recognized: hello, Confidence: 100 2012-05-15 17:01:17.648543 [INFO] switch_cpp.cpp:1227 Callback with type event 2012-05-15 17:01:17.648543 [INFO] switch_cpp.cpp:1227 'Event-Name: DETECTED_SPEECH Core-UUID: 0a387574-9ee8-11e1-8fa1-43045b098f7a FreeSWITCH-Hostname: xxxxxx FreeSWITCH-Switchname: xxxxxx FreeSWITCH-IPv4: 199.15.96.29 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-05-15%2017%3A01%3A17 Event-Date-GMT: Wed,%2016%20May%202012%2000%3A01%3A17%20GMT Event-Date-Timestamp: 1337126477648543 Event-Calling-File: switch_ivr_async.c Event-Calling-Function: speech_thread Event-Calling-Line-Number: 3549 Event-Sequence: 1209 Speech-Type: detected-speech Content-Length: 172 <?xml version="1.0"?> <result grammar="xxxxxx"> <interpretation grammar="xxxxxx" confidence="100"> <input mode="speech">hello</input> </interpretation> </result> ' EXECUTE sofia/xxxxxx/2143909113 detect_speech(pause)
tl;dr; 3 opportunities, none are urgent. virtualization isn't a need-to-have but it would be nice to know what is do-able these days... i've had the best part of 4 years away from 100% voip, and spent 2 years in academia on gentoo plus haven't worked for nearly 12 months just faffing about with J2EE stuff and a bit of recent asterisk/kamailio and now freeswitch. Hi Dario - I am creating a load test at the moment for a queue of 90 calls and comparing FS with asterisk for queuing - just reading over my old notes for SIPp. I am very familiar with asterisk through to 1.4 but have been doing plain old web sys admin with gentoo in academia for 2 years and stayed away from voip and the industry - but times/opportunities are changing up to get back into it. I have lurked in the FS irc channel for about 6 months whilst doing this other stuff and now is the time to pull the finger out so to speak. Current opportunities are 2 potential network providers and a start up i'm the tech guy for: A) This 90 call queue scenario will not be virtualised as it probably would not suit queuing which I understand to be a bit hungry for resources - it will be a failover set up for an Avaya where the avaya is on-site and this hosted-queue will be at the DC. B) I am working on another startup that does purely conferencing and sms (hence my persistence with mod_gsmopen over the last couple of days) C) Am getting some information on a hosted pabx environment where the current set up is 2 asterisk boxes with 700 registrations and about 60 concurrent calls and where to go with it. the people that run that show haven't told me too much - i.e. we want to give access to "IT Guys" to administer their own customers, we just want to bill the minutes VS they have their own hosted apps and want to keep everything 100% theirs. The great thing with (C) is that I will be doing system engineering and no support. I will read through the links you sent. I am unfamiliar with virtualisation in terms of real world usage / performance etc. but have dabbled in a convenience-for-development xen and virtualbox. Using openvz in a "you can manage your own customer handsets" is appealing - i just had a quick read of it all. But yes -- it's a load testing thing at the end of the day, and there can be no blips, or other voiping artifacts (i.e. digitized voice and the like) In general: 1. No transcoding (ever) - it will all be alaw all the way. 2. All servers have Xeon CPUs. Whether they are recent or from the last 5 years is another story depending on the availability of servers and/or cash. I think there are more servers than cash to spend - rackspace is not a problem. That being said, I just priced up some dells and supermicros to have some kind of figures ready. 3. Currently, there are 60 concurrent calls across about 700 registrations. Switches and routers are none of my concern, but they use a range of some cisco switches, a redback router, extremenetworks switch and some juniper firewalls will be in before end of year. There is no rush on any of the above, it's all early days, plus I still have some of this non-computer project work I've been doing - manual labour upgrades...fun fun fun!!! Thanks for your links and comments - I look forward to hearing more. On 15/05/2012, at 10:44 PM, Saugort Dario Garcia Tovar wrote:
Hi, Chris
That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic? - ¿How many calls have you handle in the real world? - ¿Transcoding? - ¿Have you tested your VM?
Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!!
Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware: http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.ht... http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf
On 5/15/2012 4:42 AM, Chris Mylonas wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12
-- Atentamente, Dario García Consultor.
CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I currently run a testing setup on rackspace cloud, doesn't seem to be too bad, but then again the call volume is low. On 15 May 2012 14:43, Chris Mylonas <chris@opencsta.org> wrote:
tl;dr; 3 opportunities, none are urgent. virtualization isn't a need-to-have but it would be nice to know what is do-able these days... i've had the best part of 4 years away from 100% voip, and spent 2 years in academia on gentoo plus haven't worked for nearly 12 months just faffing about with J2EE stuff and a bit of recent asterisk/kamailio and now freeswitch.
Hi Dario - I am creating a load test at the moment for a queue of 90 calls and comparing FS with asterisk for queuing - just reading over my old notes for SIPp. I am very familiar with asterisk through to 1.4 but have been doing plain old web sys admin with gentoo in academia for 2 years and stayed away from voip and the industry - but times/opportunities are changing up to get back into it. I have lurked in the FS irc channel for about 6 months whilst doing this other stuff and now is the time to pull the finger out so to speak.
Current opportunities are 2 potential network providers and a start up i'm the tech guy for: A) This 90 call queue scenario will not be virtualised as it probably would not suit queuing which I understand to be a bit hungry for resources - it will be a failover set up for an Avaya where the avaya is on-site and this hosted-queue will be at the DC.
B) I am working on another startup that does purely conferencing and sms (hence my persistence with mod_gsmopen over the last couple of days)
C) Am getting some information on a hosted pabx environment where the current set up is 2 asterisk boxes with 700 registrations and about 60 concurrent calls and where to go with it. the people that run that show haven't told me too much - i.e. we want to give access to "IT Guys" to administer their own customers, we just want to bill the minutes VS they have their own hosted apps and want to keep everything 100% theirs.
The great thing with (C) is that I will be doing system engineering and no support.
I will read through the links you sent. I am unfamiliar with virtualisation in terms of real world usage / performance etc. but have dabbled in a convenience-for-development xen and virtualbox. Using openvz in a "you can manage your own customer handsets" is appealing - i just had a quick read of it all.
But yes -- it's a load testing thing at the end of the day, and there can be no blips, or other voiping artifacts (i.e. digitized voice and the like)
In general:
1. No transcoding (ever) - it will all be alaw all the way. 2. All servers have Xeon CPUs. Whether they are recent or from the last 5 years is another story depending on the availability of servers and/or cash. I think there are more servers than cash to spend - rackspace is not a problem. That being said, I just priced up some dells and supermicros to have some kind of figures ready. 3. Currently, there are 60 concurrent calls across about 700 registrations.
Switches and routers are none of my concern, but they use a range of some cisco switches, a redback router, extremenetworks switch and some juniper firewalls will be in before end of year.
There is no rush on any of the above, it's all early days, plus I still have some of this non-computer project work I've been doing - manual labour upgrades...fun fun fun!!!
Thanks for your links and comments - I look forward to hearing more.
On 15/05/2012, at 10:44 PM, Saugort Dario Garcia Tovar wrote:
Hi, Chris
That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic? - ¿How many calls have you handle in the real world? - ¿Transcoding? - ¿Have you tested your VM?
Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!!
Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware:
http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.ht... http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf
On 5/15/2012 4:42 AM, Chris Mylonas wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12
-- Atentamente, *Dario García* Consultor.
CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info@cassidywebservices.co.uk> *03300 100 960 *F<info@cassidywebservices.co.uk> *03300 100 961 *E <info@cassidywebservices.co.uk> *andrew@cassidywebservices.co.uk *W <info@cassidywebservices.co.uk> *www.cassidywebservices.co.uk
What volume are you able to do on Rackspace? And, if I could ask, what service tier/# of cores/OS are you using there? Thanks! From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Andrew Cassidy Sent: Tuesday, May 15, 2012 10:48 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] 2012 VoIP With Virtualization I currently run a testing setup on rackspace cloud, doesn't seem to be too bad, but then again the call volume is low. On 15 May 2012 14:43, Chris Mylonas <chris@opencsta.org<mailto:chris@opencsta.org>> wrote: tl;dr; 3 opportunities, none are urgent. virtualization isn't a need-to-have but it would be nice to know what is do-able these days... i've had the best part of 4 years away from 100% voip, and spent 2 years in academia on gentoo plus haven't worked for nearly 12 months just faffing about with J2EE stuff and a bit of recent asterisk/kamailio and now freeswitch. Hi Dario - I am creating a load test at the moment for a queue of 90 calls and comparing FS with asterisk for queuing - just reading over my old notes for SIPp. I am very familiar with asterisk through to 1.4 but have been doing plain old web sys admin with gentoo in academia for 2 years and stayed away from voip and the industry - but times/opportunities are changing up to get back into it. I have lurked in the FS irc channel for about 6 months whilst doing this other stuff and now is the time to pull the finger out so to speak. Current opportunities are 2 potential network providers and a start up i'm the tech guy for: A) This 90 call queue scenario will not be virtualised as it probably would not suit queuing which I understand to be a bit hungry for resources - it will be a failover set up for an Avaya where the avaya is on-site and this hosted-queue will be at the DC. B) I am working on another startup that does purely conferencing and sms (hence my persistence with mod_gsmopen over the last couple of days) C) Am getting some information on a hosted pabx environment where the current set up is 2 asterisk boxes with 700 registrations and about 60 concurrent calls and where to go with it. the people that run that show haven't told me too much - i.e. we want to give access to "IT Guys" to administer their own customers, we just want to bill the minutes VS they have their own hosted apps and want to keep everything 100% theirs. The great thing with (C) is that I will be doing system engineering and no support. I will read through the links you sent. I am unfamiliar with virtualisation in terms of real world usage / performance etc. but have dabbled in a convenience-for-development xen and virtualbox. Using openvz in a "you can manage your own customer handsets" is appealing - i just had a quick read of it all. But yes -- it's a load testing thing at the end of the day, and there can be no blips, or other voiping artifacts (i.e. digitized voice and the like) In general: 1. No transcoding (ever) - it will all be alaw all the way. 2. All servers have Xeon CPUs. Whether they are recent or from the last 5 years is another story depending on the availability of servers and/or cash. I think there are more servers than cash to spend - rackspace is not a problem. That being said, I just priced up some dells and supermicros to have some kind of figures ready. 3. Currently, there are 60 concurrent calls across about 700 registrations. Switches and routers are none of my concern, but they use a range of some cisco switches, a redback router, extremenetworks switch and some juniper firewalls will be in before end of year. There is no rush on any of the above, it's all early days, plus I still have some of this non-computer project work I've been doing - manual labour upgrades...fun fun fun!!! Thanks for your links and comments - I look forward to hearing more. On 15/05/2012, at 10:44 PM, Saugort Dario Garcia Tovar wrote: Hi, Chris That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ¿Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ¿How you set your VM? cpu, cpu cores, memory, disk, #nic? - ¿How many calls have you handle in the real world? - ¿Transcoding? - ¿Have you tested your VM? Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!! Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware: http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.ht... http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf On 5/15/2012 4:42 AM, Chris Mylonas wrote: Hi FS Users, What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA. If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> ----- No virus found in this message. Checked by AVG - www.avg.com<http://www.avg.com/> Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 -- Atentamente, Dario García Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842<tel:%2B58%20212%209081842> Cel: +58 412 2221515<tel:%2B58%20412%202221515> dgarcia@anew.com.ve<mailto:dgarcia@anew.com.ve> http://www.anew.com.ve<http://www.anew.com.ve/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director [http://www.cassidywebservices.co.uk/media/emailsig.png] T<mailto:info@cassidywebservices.co.uk> 03300 100 960 F<mailto:info@cassidywebservices.co.uk> 03300 100 961 E<mailto:info@cassidywebservices.co.uk> andrew@cassidywebservices.co.uk<mailto:andrew@cassidywebservices.co.uk> W<mailto:info@cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk>
I have Freeswitch running on a KVM based VM right now, and it doesn't do too hot. Voicemail, MoH, and IVR prompts are choppy often but call media proxy (no transcode) works okay, though long calls start to deteriorate after about 30 minutes. I haven't tried on Xen since I bailed to KVM to get to a later kernel in 2010 when blktap2 progress was unapparent. However Xen has always been much better at I/O in my opinion. Rackspace runs Xen for their cloud servers service, and people on this list seem to report good results with Freeswitch on Rackspace cloud. Most recently I have been testing on LXC guests on Ubuntu 12.04. This has worked out as well as LXC doesn't do any hardware virtualization. Instead it is a method of isolating execution of processes in different namespaces and all guests share the host kernel. I have been migrating all of my KVM guests to LXC, and my Freeswitch install is on the list. With LXC you can set memory and CPU usage restrictions, so in my experience provides all the same benefits of full virt minus the ability to run Windows. OpenVZ is also based on process isolation rather than hardware virtualization, and would probably have the same benefits over full virt as LXC with Freeswitch. -- Thaddeus On 5/15/2012 4:12 AM, Chris Mylonas wrote:
Hi FS Users,
What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA.
If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node...
Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan <thaddeus@thogan.com> wrote:
I have Freeswitch running on a KVM based VM right now, and it doesn't do too hot. Voicemail, MoH, and IVR prompts are choppy often but call media proxy (no transcode) works okay, though long calls start to deteriorate after about 30 minutes.
I've used KVM with good results... what kind of load are we talking about? Gabe
I'm on the 512MB RAM tier. Occasionally FreeSWITCH craps itself and crashes leaving no logs but that's like every few weeks. I currently have my work calls routed through it, and take a fair few of them (one at a time mind, but I have call waiting set up on my phone so can have 2 sip sessions at once. Voicemail seems fine, and I have a conference call set up but it needs further testing. tel:+443300100002, sip:03300100002@sip.v-pbx.net:5080 or ISN 1234*1432 will get you into the conference which is probably the best test of the media capabilities. I've not done a proper load test on it either, I've mainly been using it to build a front end. I've been told by customers that my voice breaks up a little, but since setting vad=none I've seen better results. Incoming audio seems to have no issues, mind so it could just be my home internet connection which today was upgraded to 30Mbps down 2Mbps up so I may see an improvement yet. I also installed voipmonitor yesterday, so will take a look at the logs later. On 16 May 2012 06:53, Gabriel Gunderson <gabe@gundy.org> wrote:
On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan <thaddeus@thogan.com> wrote:
I have Freeswitch running on a KVM based VM right now, and it doesn't do too hot. Voicemail, MoH, and IVR prompts are choppy often but call media proxy (no transcode) works okay, though long calls start to deteriorate after about 30 minutes.
I've used KVM with good results... what kind of load are we talking about?
Gabe
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info@cassidywebservices.co.uk> *03300 100 960 *F<info@cassidywebservices.co.uk> *03300 100 961 *E <info@cassidywebservices.co.uk> *andrew@cassidywebservices.co.uk *W <info@cassidywebservices.co.uk> *www.cassidywebservices.co.uk
When it's not office hours, why don't you sent a trunk in from another box and pound the sh.. out of it to see where it breaks? On 17/05/2012, at 1:58 AM, Andrew Cassidy wrote:
I'm on the 512MB RAM tier. Occasionally FreeSWITCH craps itself and crashes leaving no logs but that's like every few weeks. I currently have my work calls routed through it, and take a fair few of them (one at a time mind, but I have call waiting set up on my phone so can have 2 sip sessions at once. Voicemail seems fine, and I have a conference call set up but it needs further testing.
tel:+443300100002, sip:03300100002@sip.v-pbx.net:5080 or ISN 1234*1432 will get you into the conference which is probably the best test of the media capabilities.
I've not done a proper load test on it either, I've mainly been using it to build a front end.
I've been told by customers that my voice breaks up a little, but since setting vad=none I've seen better results. Incoming audio seems to have no issues, mind so it could just be my home internet connection which today was upgraded to 30Mbps down 2Mbps up so I may see an improvement yet.
I also installed voipmonitor yesterday, so will take a look at the logs later.
On 16 May 2012 06:53, Gabriel Gunderson <gabe@gundy.org> wrote: On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan <thaddeus@thogan.com> wrote:
I have Freeswitch running on a KVM based VM right now, and it doesn't do too hot. Voicemail, MoH, and IVR prompts are choppy often but call media proxy (no transcode) works okay, though long calls start to deteriorate after about 30 minutes.
I've used KVM with good results... what kind of load are we talking about?
Gabe
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director
T 03300 100 960 F 03300 100 961 E andrew@cassidywebservices.co.uk W www.cassidywebservices.co.uk
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Check out http://wiki.opencsta.org/index.php/SIPp the registration scenario, or just set up IP based ACL without auth to send you a call every 10 seconds or so in a queue. (or set up another box and some sort of call generation mechanism) - I'm currently using SIPp to register and bailing out of executing after the first registration, then listening for any media and echoing it back for queues. They are very obedient agents :) I'm sure something similar could be done for the conference, just bridge one conference channel to a file - the SIPp buggers will echo it back! You'd be best putting some monitoring software on it too (zabbix.com is pretty good - might be a bit of a learning curve, but if you wanted to put the agent/proxy on there and route it back to my install i can give you a login - it's phoning home back to the zabbix server is over ssl tunnels) Maybe set debug on as well, hopefully that tells you a bit more. On 17/05/2012, at 1:58 AM, Andrew Cassidy wrote:
I'm on the 512MB RAM tier. Occasionally FreeSWITCH craps itself and crashes leaving no logs but that's like every few weeks. I currently have my work calls routed through it, and take a fair few of them (one at a time mind, but I have call waiting set up on my phone so can have 2 sip sessions at once. Voicemail seems fine, and I have a conference call set up but it needs further testing.
tel:+443300100002, sip:03300100002@sip.v-pbx.net:5080 or ISN 1234*1432 will get you into the conference which is probably the best test of the media capabilities.
I've not done a proper load test on it either, I've mainly been using it to build a front end.
I've been told by customers that my voice breaks up a little, but since setting vad=none I've seen better results. Incoming audio seems to have no issues, mind so it could just be my home internet connection which today was upgraded to 30Mbps down 2Mbps up so I may see an improvement yet.
I also installed voipmonitor yesterday, so will take a look at the logs later.
On 16 May 2012 06:53, Gabriel Gunderson <gabe@gundy.org> wrote: On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan <thaddeus@thogan.com> wrote:
I have Freeswitch running on a KVM based VM right now, and it doesn't do too hot. Voicemail, MoH, and IVR prompts are choppy often but call media proxy (no transcode) works okay, though long calls start to deteriorate after about 30 minutes.
I've used KVM with good results... what kind of load are we talking about?
Gabe
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director
T 03300 100 960 F 03300 100 961 E andrew@cassidywebservices.co.uk W www.cassidywebservices.co.uk
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Not much on this particular box. 20 registered clients, all behind NAT and from all over the US. Average of about 5 concurrent calls. Host is an Opteron 6100 series 8 core w/16 GB RAM running Ubuntu 10.04, guest has 4 GB RAM and exclusive access to 2 cores on the host. I figured the issue was KVM because the same setup on the same hardware configuration with an LXC guest is handles a testing load of similar proportions without issue. Everything here is moving to LXC over the next three months anyway. My observations with KVM must be considered under the premise that I have not identified any specific cause for the issues related to KVM. Also I have not run Freeswitch on Ubuntu 10.04 on bare metal. -- Thaddeus On 5/16/2012 12:53 AM, Gabriel Gunderson wrote:
On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan<thaddeus@thogan.com> wrote:
I have Freeswitch running on a KVM based VM right now, and it doesn't do too hot. Voicemail, MoH, and IVR prompts are choppy often but call media proxy (no transcode) works okay, though long calls start to deteriorate after about 30 minutes.
I've used KVM with good results... what kind of load are we talking about?
Gabe
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (10)
-
Andrew Cassidy
-
Brett Clark - Grasshopper
-
Chris Mylonas
-
Gabriel Gunderson
-
Giovanni Maruzzelli
-
jay binks
-
Noel Morgan
-
Saugort Dario Garcia Tovar
-
Thaddeus Hogan
-
Wesley Akio