[Freeswitch-users] Audio quality issues

Wed Oct 23 01:39:50 MSD 2013

You have turned off CDR, but what about logging?  Logging at certain levels
gets pretty taxing at these call rates.
Have you implemented all the hints from
https://wiki.freeswitch.org/wiki/Performance_Tuning and
https://wiki.freeswitch.org/wiki/Performance_testing_and_configurations ?

Guillermo

On Tue, Oct 22, 2013 at 3:41 PM, Deon Vermeulen <vermeulen.deon at gmail.com>wrote:

> My 2cents is to move to bare metal.
> We ran a Guest on Commercial Licensed ESX and without real load (>50%)
> started seeing bad behavior.
> We moved to Bare Metal just over 3weeks ago and since then no more strange
> behavior.
>
> I've setup OpenVZ for LAB environment and so far no issues with almost the
> same load when using Commercial Licensed ESX.
>
>
> Kind Regards
>
> Sent from my iPhone
>
> > On Oct 22, 2013, at 19:28, "Miha" <miha at softnet.si> wrote:
> >
> > Hi,
> >
> >
> > I would be also interested in this as i am experiancing
> > same issue, poor audio quality and media dropping.
> >
> > Br,
> >
> > Miha
> >
> > On Tue, 22 Oct 2013 16:03:29 +0200
> > Grant Bagdasarian <GB at cm.nl> wrote:
> >> Yes, it does!
> >>
> >> I also found this:
> > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo
> >>
> >> Echo application for FS.
> >> Going to see if that works first. If not, I'll setup a
> >> Sipp in server mode.
> >>
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
> >> Behalf Of Ali Pey
> >> Sent: Tuesday, October 22, 2013 3:50 PM
> >> To: FreeSWITCH Users Help
> >> Subject: Re: [Freeswitch-users] Audio quality issues
> >>
> >> You can make calls from sipp that also terminates on sipp
> >> and then play a wave file in sipp.
> >>
> >> Change your dial plan in FS to route the calls to an
> >> instant of sipp that can terminate the calls.
> >>
> >> Does this make sense?
> >>
> >> On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian
> >> <GB at cm.nl<mailto:GB at cm.nl>> wrote:
> >> A while back I used iotop to measure the disk access, and
> >> FS was hardly using any io during tests.
> >>
> >> How do I simulate two way audio?
> >> I know I can make Sipp send an RTP stream using a pcap
> >> file, but how do I make FS sent RTP back which is not
> >> read from disk? Does FS have an echo application?
> >> Or is it enough for Sipp to send the media?
> >>
> >> From:
> > freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org>
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org>]
> >> On Behalf Of Ali Pey
> >> Sent: Tuesday, October 22, 2013 2:25 PM
> >>
> >> To: FreeSWITCH Users Help
> >> Subject: Re: [Freeswitch-users] Audio quality issues
> >>
> >> I think the problem here is that you are playing a file
> >> for every call for the duration of the call. The
> >> bottleneck seems to be the disk access. If there were to
> >> be two way audio path, FS would only proxy the media
> >> which would be quite faster as there would be no file
> >> reading and playing involved. Attempt a test case with
> >> fewer or no file play and only media proxy and test
> >> again.
> >>
> >>
> >> On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian
> >> <GB at cm.nl<mailto:GB at cm.nl>> wrote:
> >> The network shouldn't be an issue, since we have at least
> >> 1Gbps lines. The tests stay within the network.
> >>
> >> I forgot to mention the calls are being distributed
> >> across two machines by a Kamailio instance.
> >> So for a total of 800 concurrent calls generated by Sipp,
> >> each machine has 400 active calls.
> >> CPU load reaches about 70% per machine.
> >>
> >> At this point both FS machines are virtualized, since the
> >> performance gain wasn't that much compared to physical.
> >> The VM host shows it is using ~3/4 of its CPU resources.
> >>
> >> Htop shows that the normal priority threads(green) and
> >> the kernel threads(red) are about the same length.
> >>
> >> Also, FS is running on Ubuntu Server 12.04 x64.
> >>
> >> From:
> > freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org>
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org>]
> >> On Behalf Of Stanislav Sinyagin
> >> Sent: Tuesday, October 22, 2013 11:11 AM
> >> To: FreeSWITCH Users Help
> >> Subject: Re: [Freeswitch-users] Audio quality issues
> >>
> >> 800 calls at 64kbps is 51Mbps.
> >> Could there be a network issue, like a 100Mbps line
> >> between the endpoints?
> >>
> >> How heavy is your CPU load?  "htop" command would be
> >> helpful in this.
> >>
> >>
> >> ________________________________
> >> From: Grant Bagdasarian <GB at cm.nl<mailto:GB at cm.nl>>
> >> To: "FreeSWITCH Users Help
> > (freeswitch-users at lists.freeswitch.org<mailto:
> freeswitch-users at lists.freeswitch.org>)"
> > <freeswitch-users at lists.freeswitch.org<mailto:
> freeswitch-users at lists.freeswitch.org>>
> >> Sent: Tuesday, October 22, 2013 10:14 AM
> >> Subject: [Freeswitch-users] Audio quality issues
> >>
> >> Hello,
> >>
> >> I was wondering what the maximum concurrent calls for FS
> >> before audio quality becomes an issue? I assume the specs
> >> of the machine would also affect this.
> >> We are currently running FS on a Six Core (12 Threads)
> >> Intel E5-2430 CPU and get about 800 concurrent calls at
> >> 10-20 CPS. The audio quality at these rates is still
> >> fair, but we do notice some quality issue's.
> >> Going above these numbers screws up the audio quality:
> >> choppy sound, audio drops etc. We aren't doing any heavy
> >> media processing, just simply playing a file (G711-Alaw)
> >> which lasts about 2 minutes during the load test.
> >> These numbers are for one way audio, where Sipp doesn't
> >> echo the RTP back. These numbers get lower once Sipp
> >> echo's the RTP.
> >>
> >> I've tried FS on a physical box and also on a virtual box
> >> (ESXi 5.1), but the performance gain on physical vs
> >> virtual isn't that much.
> >>
> >> I disabled all the modules we don't need, like CDR's,
> >> conferencing, etc.
> >>
> >> Are there any parameters(config files)/modules that can
> >> affect the quality of the audio stream?
> >>
> >> Regards,
> >>
> >> Grant
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> >> Server
> >> 
> >>
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> >> [cid:~WRD000.jpg]
> > _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> >> http://www.freeswitchsolutions.com
> >>
> >> FreeSWITCH-powered IP PBX: The CudaTel Communication
> >> Server
> >> 
> >>
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> >
> >
> > _________________________________________________________________________
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> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
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>
> 
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-- 
Guillermo Ruiz Camauer
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