[Freeswitch-users] Choppy audio when conferencing 4+ participants

Stanislav Sinyagin ssinyagin at gmail.com
Wed Apr 19 00:56:31 MSD 2017

as discussed many times already, there are several factors that affect
audio quality in conferences or playback of audio files (regular
bridged calls are less sensitive to this).

1) Less often than #2, but sometimes you don't get the clock that is
precise enough. Conferencing and audio playback require a good
reference clock, and some virtualized environments don't provide it.
As noted somewhere in the wiki, by default AWS provides a kernel with
really bad clock frequency, so you ought to change the kernel.

2) Any public VM hosting is a best-effort service, and you are very
likely to have noisy neighbors. They all produce spikes in CPU load,
and your VM is by no means prioritized against the others. So, your FS
needs to compose an RTP frame and send it on-time, but it simply
doesn't get the CPU cycles when it needs them. This leads to rather
high jitter, and it's totally unpredictable. It may also depend on
time of day or day of week, and you get totally different quality
measurements with 5 minutes interval.

So, for any kind of IVR or conferencing server, you ought to have
physical hardware, or a hypervisor under your control, so that you
allocate a guaranteed CPU resource to your FreeSWITCH machine.

There's actually a quite broad choice of physical hardware hosting, so
you can place your media servers on physical boxes, while leaving your
application logic at your beloved AWS. The physical hardware doesn't
have to be big: without transcoding, an average Celeron server can
handle dozens of conference legs without problems. It's also quite
easy to estimate your CPU capacity in advance, before sending
production traffic: you create a conference, one leg goes to your ear,
then few dozens of legs go to a silent sink on another server, and one
leg is playing back some reference audio. Then you increase the number
of participants until you start hearing quality impairments.

There are also commercial tools for measuring voice quality, if you
intend to run quality checks periodically.


On Fri, Apr 14, 2017 at 6:48 PM, Bilal Dar <bilal at rgate-systems.com> wrote:
> I have been struggling with an issue for almost 2 weeks.
> Our regular calls have no quality issue and looking RTCP statistics network
> conditions are perfect. We have normally on peak hr 60 calls and around 10
> conferences.
> We have noticed that when we have 2 conferences of 4 or 5 participants,
> audio starts breaking for the users who are on conference. Regular calls do
> not experience any quality degradation.
> I upgraded the server to specs of 30Gig memory and 8 vCPU but still the
> issue exists. Common thing I have noticed even during off-peak hrs is that
> two 4+ participant call can cause the issue.
> I have ruled out network & hardware. Last change I made was moved all users
> to G.711 from G.722. Now I am not sure what other steps I can take.
> Appreciate any suggestions.
> Thanks
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