[Freeswitch-users] Improving voice quality

[Adam Ben-Ayoun]

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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