Hello: I set up a FreeSwitch-based VoIP server as a host on my cluster, and am having audio problems. I'm not 100% sure if its virtualization related or network related yet. But I would like to optimize my VM for VoIP (or rather, tell Ovirt all the "right settings" to optimize that VM to VoIP). Does anyone have any specific suggestions? Are there known issues with VoIP on Ovirt-managed clusters? (I know well reputed companies that sell VoIP server virtual hosting and guarantee the performance, so I know VoIP Virtualization is possible, just need to know if its recommended with Ovirt, and if so what do I need to do to give it the best chance of success?) Thanks! --Jim
Once upon a time, Jim Kusznir <jim@palousetech.com> said:
Are there known issues with VoIP on Ovirt-managed clusters? (I know well reputed companies that sell VoIP server virtual hosting and guarantee the performance, so I know VoIP Virtualization is possible, just need to know if its recommended with Ovirt, and if so what do I need to do to give it the best chance of success?)
I am running Asteria (an Asterisk-based PBX system targeted at small call-center type setups) in an oVirt VM with no problems. We typically have 30-50 calls at a time during the business day. I've also set up Digium's Switchvox in an oVirt VM without issue (small office setup, so not a lot of simultaneous calls). -- Chris Adams <cma@cmadams.net>
Hi, Can you please describe the application network requirements? Does it relay on low latency? Pass-through or SR-IOV could help with reducing that. Yaniv Dary Technical Product Manager Red Hat Israel Ltd. 34 Jerusalem Road Building A, 4th floor Ra'anana, Israel 4350109 Tel : +972 (9) 7692306 8272306 Email: ydary@redhat.com IRC : ydary On Wed, Jan 4, 2017 at 6:03 AM, Chris Adams <cma@cmadams.net> wrote:
Once upon a time, Jim Kusznir <jim@palousetech.com> said:
Are there known issues with VoIP on Ovirt-managed clusters? (I know well reputed companies that sell VoIP server virtual hosting and guarantee the performance, so I know VoIP Virtualization is possible, just need to know if its recommended with Ovirt, and if so what do I need to do to give it the best chance of success?)
I am running Asteria (an Asterisk-based PBX system targeted at small call-center type setups) in an oVirt VM with no problems. We typically have 30-50 calls at a time during the business day.
I've also set up Digium's Switchvox in an oVirt VM without issue (small office setup, so not a lot of simultaneous calls).
-- Chris Adams <cma@cmadams.net> _______________________________________________ Users mailing list Users@ovirt.org http://lists.ovirt.org/mailman/listinfo/users
Once upon a time, Yaniv Dary <ydary@redhat.com> said:
Can you please describe the application network requirements? Does it relay on low latency? Pass-through or SR-IOV could help with reducing that.
For VoIP, latency can be an issue, but the amount of latency from adding VM networking overhead isn't a big deal (because other network latency will have a larger impact). 10ms isn't really a problem for VoIP for example. The bigger network concern for VoIP is jitter; for that, the only solution is to not over-provision hardware CPUs or total network bandwidth. -- Chris Adams <cma@cmadams.net>
Sorry for the delayed response, I finally found where gmail hid this response... :( So the application is FusionPBX, a FreeSwitch-based VoIP system, running on a very unloaded (1% cpu load, 2-4 VMs running) system. I've been experiencing intermittent call breakup, for which external support immediately blamed on the virtualization solution claiming that "You can't virtualize VoIP systems without causing voice breakup and other call quality issues". Previously, I had attempted to run FreePBX (asterisk-based) on a Hyper-V system, and I did find that to be the case; moving over to very weak, but dedicated hardware, fixed the problem immediately. Since I sent this message, I did extensive testing with my system, and it appears that the breakup is in fact network related. I've been able to do phone to phone calls on the local network for extended durations without issue, and even have phone to phone calls on external networks without issue. However, calls going to my VoIP provider do break up, so it appears to be the network route to my provider. So, oVirt does not appear to be to blame (which I didn't think so, but was hoping for some "expert information" to support this...It appears that I got that and more with my tests). Thank you again for your work on such a great product! --Jim On Wed, Jan 4, 2017 at 10:08 AM, Chris Adams <cma@cmadams.net> wrote:
Once upon a time, Yaniv Dary <ydary@redhat.com> said:
Can you please describe the application network requirements? Does it relay on low latency? Pass-through or SR-IOV could help with reducing that.
For VoIP, latency can be an issue, but the amount of latency from adding VM networking overhead isn't a big deal (because other network latency will have a larger impact). 10ms isn't really a problem for VoIP for example.
The bigger network concern for VoIP is jitter; for that, the only solution is to not over-provision hardware CPUs or total network bandwidth.
-- Chris Adams <cma@cmadams.net> _______________________________________________ Users mailing list Users@ovirt.org http://lists.ovirt.org/mailman/listinfo/users
On Feb 11, 2017 7:58 AM, "Jim Kusznir" <jim@palousetech.com> wrote: Sorry for the delayed response, I finally found where gmail hid this response... :( So the application is FusionPBX, a FreeSwitch-based VoIP system, running on a very unloaded (1% cpu load, 2-4 VMs running) system. I've been experiencing intermittent call breakup, for which external support immediately blamed on the virtualization solution claiming that "You can't virtualize VoIP systems without causing voice breakup and other call quality issues". Previously, I had attempted to run FreePBX (asterisk-based) on a Hyper-V system, and I did find that to be the case; moving over to very weak, but dedicated hardware, fixed the problem immediately. Since I sent this message, I did extensive testing with my system, and it appears that the breakup is in fact network related. I've been able to do phone to phone calls on the local network for extended durations without issue, and even have phone to phone calls on external networks without issue. However, calls going to my VoIP provider do break up, so it appears to be the network route to my provider. So, oVirt does not appear to be to blame (which I didn't think so, but was hoping for some "expert information" to support this...It appears that I got that and more with my tests). Great to hear. I do believe that setting affinity and possibly taking into account NUMA makes sense. Perhaps using SR-IOV is needed for low latency. There is interesting work upstream qemu to improve throughout and reduce latency in the expanse of more CPU usage. Lastly, real time, mainly kernel and qemu-kvm, is also technology that might be needed for some workloads. See [1]. Y. [1] https://mpolednik.github.io/2016/09/19/real-time-host-in-ovirt/ Thank you again for your work on such a great product! --Jim On Wed, Jan 4, 2017 at 10:08 AM, Chris Adams <cma@cmadams.net> wrote:
Once upon a time, Yaniv Dary <ydary@redhat.com> said:
Can you please describe the application network requirements? Does it relay on low latency? Pass-through or SR-IOV could help with reducing that.
For VoIP, latency can be an issue, but the amount of latency from adding VM networking overhead isn't a big deal (because other network latency will have a larger impact). 10ms isn't really a problem for VoIP for example.
The bigger network concern for VoIP is jitter; for that, the only solution is to not over-provision hardware CPUs or total network bandwidth.
-- Chris Adams <cma@cmadams.net> _______________________________________________ Users mailing list Users@ovirt.org http://lists.ovirt.org/mailman/listinfo/users
_______________________________________________ Users mailing list Users@ovirt.org http://lists.ovirt.org/mailman/listinfo/users
participants (4)
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Chris Adams -
Jim Kusznir -
Yaniv Dary -
Yaniv Kaul