Hello all, let me also give some experience from the VirtualBox side (Community Version). Host machine ========== AMD X2 64 3800 with 8GB of RAM OS is a generic Debian 4.0R5 with Kernel 2.6.18-6-amd64 No special parameters in the Kernel. Started with VirtualBox 1.5 and now on 2.0.x Client machine (freeswitch) =========== Ubuntu 8.041 Generic Kernel 2.6.24-18-generic #1 SMP, 1 CPU Experience: =========== A single call produces about 20% CPU load. So this is not usefull for any production environment. I did not discover any dropouts in a normal call between internal and external UAs/gateways since 6 months. So for testing purposes its fine. Voice between User Agents is always fine. Seldomly I hear choppy voice when announcements are played. After some minutes these problems go away. Resume =========== For testing/development purposes, FS on VirtualBox is fine. For any productive environment it's not really usable in our environment. Comparison with Asterisk ================= Asterisk never worked in this environment: Choppe voice between UAs and when playing sound. 100% CPU load on a single call. ========== Best regards Peter Gregory Boehnlein schrieb:
You'll never fix this. Voice is a latency specific application unless you figure out how to manipulate time. Any virtualization platform is going to provide less timing granularity than raw hardware.
Hello Ken, hello all,
I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread.
I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system.
Did you experience the same and/or do you have pointers on how to troubleshoot and fix this?
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