It all depends really. On some solutions (depending on hypervisor, hardware and guest OS) the main problem can be that 20ms is a too short period to sleep, in that case a larger frame (ie 40ms) might help out. But mostly the problem is that the clock is not accurate enough, and fluxtuates from time to time (sometimes 20ms is 10, next round 30 etc). After my latest testing I've found out that latest ESXi (4.1) works quite good, as long as you make sure not to share out more CPU's to the guests then you have actual cores on the hardware, and makes sure to use a separate NIC for that guest OS. I will summarize the results from my tests on the wiki soon, but I'm away for now, and won't get access to all the information I need until I get back - after the ClueCon event(!). If using any kind of cloud environment (like the VPS the original questions was about) that's not under your own control, I highly recommend to solve it another way. Those solutions are usually to much loaded to get an accurate time. However, I do know that using for instance EC2 (xen) has worked for some, but that would also require that host to be on Linux (the original question was about Windows). /Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Bryan Smart [bryansmart@bryansmart.com] Skickat: den 28 juli 2011 13:52 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Peter, thanks for those details.
What sort of trade-offs, if any, are useful to improve audio for conferences and IVRs on VMs? Will increasing the interval setting for a conference help? What about using larger frames for codecs? I know that latency, beyond a point, is a serious quality issue of its own. Still, do you know what compromises provide the best improvement? Bryan
On 7/28/11, Peter Olsson <peter.olsson@visionutveckling.se> wrote:
I feel I need to explain my last statement a little further :)
First of all, you can run FS on virtual machines (even though I never recommend it), but you need to be in control of the virtual environment, or totally trust the ones controlling it.
There are two basic problems when running on virtual hardware - timing and network I/O. For timing to work you need to make sure that the one controlling the virtual environment NEVER give out more CPU cores then there are available on the hardware. If they give out more, and your machine need to share resources with another machine, you will be in trouble. As soon as the other virtual machine starts to load the CPU, your machine will get bad timing. Also - usually a one core VM will get better timing then a two+ core VM.
Also, to be safe on the network side, you need your own NIC, that is not shared with other machines.
If you feel you can control this, you might be successful with the setup, if you can't control this, you're either lucky (and things work anyway), or you'll get bad audio.
Also, just passing RTP for normal call isn't the most critical issue (you don't even have to use a timer there), the most critical timing issues will occur in conferences or when using a local IVR on the machine.
I've done lots of testing for this in my lab, since many of my customers wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual hosts, and I've found the above to be quite critical to succeed with your virtual implementation.
/Peter
____________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Giovanni Maruzzelli [gmaruzz@gmail.com] Skickat: den 28 juli 2011 09:37 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
Hey guys 'n gal,
the original post is asking for Windows!!!
-giovanni
On 7/28/11, Sam <lakersman2006@yahoo.com> wrote:
i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.
________________________________ From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Wednesday, July 27, 2011 3:20 PM Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!
I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels. Linode says the CPU is the most underused resource on their systems...
-Avi
On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc@freeswitch.org> wrote:
On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006@yahoo.com> wrote:
I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
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