[Freeswitch-users] Improving voice quality

[Oleg Stolyar]

Have you tried it with non-mobile WebRTC clients?  Is there the same
problem?

I am using mod_conference on FS hosted on AWS m3.xlarge instance and I have
not noticed audio problem.  But my WebRTC clients are not mobile but rather
Chrome on PCs.

On Tue, May 12, 2015 at 9:59 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
wrote:

> Hi guys,
>
> We are using Freeswitch as a audio "MCU" for WebRTC using mod_conference,
> we are currently using mobile clients on both Android and iOS. The voice
> quality is good but not as good as Hangouts for example, there are small
> cut-offs every now and then that affects the overall experience. We are
> trying to understand the reason for that, maybe it's Freeswitch mixing
> algorithm, our servers infrastructure, or something else. We did tried
> Janus audio conference demo and it was very good (we used chrome on Android
> which should use pretty much the same stack). We really need help figuring
> this out. I want to first make sure it's not our infrastructure (mainly
> network/CPU). We are using virtualized m3.large on AWS (2 vCPUs and 7.5GB
> RAM). Maybe there's a certain instance type on a certain provider that is
> known to give the best quality? Any ideas on how to go about this?
>
> Thanks,
> Adam
>
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