<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"><base href="x-msg://5590/"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Any voip running on a very heavily loaded VM host runs the risk of having audio issues. The question is, is it so heavily loaded that freeswitch will be starved of clock cycles or not. Testing it is the only way to see. Even if it is to some extent, things that don't need timing like regular bridged calls will be much more tolerant of some load than things that require more timing like playing sound files or conferences. I would test spinning up a bunch of channels and calling in and listening to a milliwatt tone to see if its clean or not.<div><br></div><div>Mike</div><div><br><div><div>On Sep 2, 2013, at 2:34 AM, Grant Bagdasarian <<a href="mailto:GB@cm.nl">GB@cm.nl</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div lang="EN-US" link="blue" vlink="purple" style="font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div class="WordSection1" style="page: WordSection1; "><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; ">Hello,<o:p></o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; ">I’ve been going through some threads about virtualization, some say it works for them, others say it doesn’t. Multiple platforms are used like MS Hyper-V, VM Ware, etc.<o:p></o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; ">We currently run FS on a physical box, which works great.<o:p></o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; ">We have our own virtual environment, VMWare vSphere 5.0. Our 5.0 environment is running under heavy load and I was wondering would FS (configured properly for correct timing) still function properly and able to handle hundreds of concurrent calls on a heavy loaded VMWare host? The FS virtual will also process RTP in default mode. I was hoping if someone had any experience using this setup and could give me some insights on this matter. The FS virtuals will function only as SBC’s, so no complex dial plans, no call recording, no conferencing. Just bridging the incoming call leg with an outgoing one.<o:p></o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; ">VMWare has a tech paper about running VoIP on their virtual machine, and they claim it works. (<a href="http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf" style="color: purple; text-decoration: underline; ">http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf</a>). In the paper they used Mitel vMCD, but I’m not sure if the software has been optimized to run on VMWare so the tests would come out better than running any other telephony software like FreeSwitch or Asterisk.<o:p></o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; ">Grant<o:p></o:p></div></div>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org" style="color: purple; text-decoration: underline; ">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com" style="color: purple; text-decoration: underline; ">http://www.freeswitchsolutions.com</a><br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br><a href="http://www.cudatel.com" style="color: purple; text-decoration: underline; ">http://www.cudatel.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org" style="color: purple; text-decoration: underline; ">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org" style="color: purple; text-decoration: underline; ">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com" style="color: purple; text-decoration: underline; ">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" style="color: purple; text-decoration: underline; ">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" style="color: purple; text-decoration: underline; ">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:http://<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" style="color: purple; text-decoration: underline; ">lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" style="color: purple; text-decoration: underline; ">http://www.freeswitch.org</a><br></div></blockquote></div><br></div></body></html>